From devel at thom.fr.eu.org Tue Dec 1 00:37:59 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Tue, 01 Dec 2009 09:37:59 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= Message-ID: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale at hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [12] iax:guest at conference.freeswitch.org/888 [13] googletalk:conf+888 at conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:FreeSWITCH-users at lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale at hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale at gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888 at conference.freeswitch.org [13] http://iax:guest at conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/06045a86/attachment.html From dome at tel.co.th Tue Dec 1 05:43:06 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 1 Dec 2009 20:43:06 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 Message-ID: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Dear All, I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and freeswitch 1.0.5 pre 7 My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and 2 GB memmory. when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% when i disable echo cancel FS can start but when first call incominng got 100% again. Someone can help me? BG Dome -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/5082ddfe/attachment.html From juanbackson at gmail.com Tue Dec 1 05:59:42 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 1 Dec 2009 21:59:42 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true Message-ID: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> Hi, I found that with bypass_media=true, freeswitch would change c= to FS's own IP. I think this is a misconfiguration. Does anyone know what config could have caused that? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/30430685/attachment.html From juanbackson at gmail.com Tue Dec 1 07:08:12 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 1 Dec 2009 23:08:12 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> Message-ID: <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> In the following trace, 102 is FS IP, 104 is calling party and 13 is called party. with bypass_media, FS still changes c=IN IP4 192.168.1.102 Any idea why? freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 22:56:33.782715: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport From: >;tag=786224322 To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 CSeq: 37 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 397 v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 From: >;tag=786224322 To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 CSeq: 37 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ PhonerLite at 192.168.1.102 entering state [received][100] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259708193783162] Caller-Channel-Created-Time: [1259708193783162] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [786224322] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_bypass_media: [true] variable_proxy_media: [true] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ 90964111 at 192.168.1.116:9390 SOFIA ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g Max-Forwards: 69 From: "PhonerLite" >;tag=8tH6Xjt2XaU9F To: Call-ID: 9d052856-596f-122d-1b98-0022190e9476 CSeq: 123735760 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 404 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/56510807/attachment-0001.html From moises.silva at gmail.com Tue Dec 1 08:32:04 2009 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 1 Dec 2009 11:32:04 -0500 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Message-ID: On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: > Dear All, > I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I > use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and > freeswitch 1.0.5 pre 7 > > My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and > 2 GB memmory. > when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% > when i disable echo cancel FS can start but when first call incominng got > 100% again. > > > Someone can help me? > > > I suggest you to start using freeswitch trunk (and openzap trunk). In the other hand, when you talk about 4E1 R2, you mean R2 as the telephony signaling or is this some kind of board brand? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1dae8f4e/attachment.html From mike at jerris.com Tue Dec 1 08:42:44 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 11:42:44 -0500 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> What is the jira bug number on this voicemail email issue? I don't recall seeing it. Mike On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine wrote: > > Are you on SVN trunk? As far as I recall the callee_id_number/name > stuff isnt in 1.0.4. > > No, because the SVN has problems with Emailing the voicemail... > > We use 1.0.4 and set sip_callee_id_number/name which works. I would > like to not set it and get it from the other side... > > Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 1 08:46:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 11:46:08 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> Message-ID: The only way this would happen would be if this is set to proxy media not bypass. Are you setting both? Mike On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > In the following trace, 102 is FS IP, 104 is calling party and 13 > is called party. > > with bypass_media, FS still changes c=IN IP4 192.168.1.102 > > Any idea why? > > > freeswitch at localhost.localdomain> recv 951 bytes from udp/ > [192.168.1.104]:5060 at 22:56:33.782715: > > --- > --------------------------------------------------------------------- > INVITE sip:90964111 at 192.168.1.102 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport > From: ;tag=786224322 > To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > Contact: > Content-Type: application/sdp > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE > Max-Forwards: 70 > Supported: 100rel, replaces > User-Agent: SIPPER for PhonerLite > Content-Length: 397 > > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendrecv > > --- > --------------------------------------------------------------------- > send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: > > --- > --------------------------------------------------------------------- > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 > From: ;tag=786224322 > To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Content-Length: 0 > > > --- > --------------------------------------------------------------------- > 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a- > ee3b-40d4-910d-3b1579f9a273] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ > internal/PhonerLite at 192.168.1.102 entering state [received][100] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > > EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() > 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259708193783162] > Caller-Channel-Created-Time: [1259708193783162] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [786224322] > variable_sofia_profile_name: [internal] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] > variable_sip_call_id: [003C8E1B-F8DC-DE11- > A853-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_bypass_media: [true] > variable_proxy_media: [true] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_endpoint_disposition: [RECEIVED_NOMEDIA] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > > > 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ > 90964111 at 192.168.1.116:9390 Patched SDP > --- > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > +++ > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send > signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 > (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ > 90964111 at 192.168.1.116:9390 SOFIA ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/ > internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send > signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to > sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > going to sleep > send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: > > --- > --------------------------------------------------------------------- > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g > Max-Forwards: 69 > From: "PhonerLite" ;tag=8tH6Xjt2XaU9F > To: > Call-ID: 9d052856-596f-122d-1b98-0022190e9476 > CSeq: 123735760 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 404 > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off > > v=0 > o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/92126985/attachment-0001.html From dome at tel.co.th Tue Dec 1 09:02:18 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Dec 2009 00:02:18 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Message-ID: <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> 2009/12/1 Moises Silva > On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: > >> Dear All, >> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). >> I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and >> freeswitch 1.0.5 pre 7 >> >> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and >> 2 GB memmory. >> when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% >> when i disable echo cancel FS can start but when first call incominng got >> 100% again. >> >> >> Someone can help me? >> >> >> > I suggest you to start using freeswitch trunk (and openzap trunk). > > I'll try > In the other hand, when you talk about 4E1 R2, you mean R2 as the telephony > signaling or is this some kind of board brand? > telephony signaling Thanks. Dome C. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/b5b2b584/attachment.html From yehavi.bourvine at gmail.com Tue Dec 1 09:11:24 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 19:11:24 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> Message-ID: It is MODAPP-373. Thanks, __yehavi: 2009/12/1 Michael Jerris > What is the jira bug number on this voicemail email issue? I don't > recall seeing it. > > Mike > > On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine > wrote: > > > > Are you on SVN trunk? As far as I recall the callee_id_number/name > > stuff isnt in 1.0.4. > > > > No, because the SVN has problems with Emailing the voicemail... > > > > We use 1.0.4 and set sip_callee_id_number/name which works. I would > > like to not set it and get it from the other side... > > > > Thanks! __Yehavi: > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/8a2b2ecb/attachment.html From msc at freeswitch.org Tue Dec 1 09:31:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 09:31:47 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: <200911291906.51520.errotan@gmail.com> References: <200911291906.51520.errotan@gmail.com> Message-ID: <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> On Sun, Nov 29, 2009 at 10:06 AM, Pusk?s Zsolt wrote: > Hi Guys! > > I'm using the latest svn (15711) with the default xml config. Only modified > cdr_csv.conf.xml the line to name="legs" > value="ab"/> > > Here is what i do: > > 1. 1000 calls 1001 (1001 answers the call) > 2. 1001 do blind transfer to 1002 (using *1) > 3. 1001 hangs up > 4. 1002 answers the call > 5. 1002 and 1000 hangs up > > 3 cdr records are generated (simplified): > > from,to,start,duration > "1000" "1001" "2009-11-29 15:21:53" "53" "50" > "1000" "1002" "2009-11-29 15:21:53" "79" "76" > "1000" "1002" "2009-11-29 15:22:46" "26" "23" > > As you can see the second cdr is incorrect because 1000 doesn't speak with > 1002 for 76 second. > > Is this a normal ? Is it possible to make only 2 record ? > > You may want to turn on mod_xml_curl and look at XML CDRs, comparing them to the corresponding CSV files. That should help you figure out why the values in the CSV are what they are. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/45551bfa/attachment.html From msc at freeswitch.org Tue Dec 1 09:38:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 09:38:42 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: Message-ID: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> On Wed, Nov 25, 2009 at 9:42 PM, Joseph L. Casale wrote: > I need to make faxing easy for some very computer illiterate folk. I am > using an email > service and going to use procmail to print anything incoming automatically > but they cant > get the hang of scanning to an email app, so I am going to buy a Linksys > PAP2T as per the > wiki. > > Since the setup will never receive inbound remote faxes, I just need to > direct all fax's > sent from the FXS port (that extension) to the email script in the wiki > substituting the > destination # as the alias portion of the email. > > So if I create a dialplan that catches the caller_id_number of the FXS > port, does the $1 > variable exist in the following scenario: > > > > In this case, the $1 will only contain whatever is in the parens in your expression, i.e. What do you have for your expression? -MC > as that's how our service requires fax's, the 10 digit # at their domain, > fax.com. Is this > a plausible setup? > > Lastly, I see in the interop list that Audiocodes Mediapack 114 is > supported, but the 202 > is not listed, is that simply because its new or is it known to not work? > Given that its > the same price as the Linksys, I would rather get it. > > Thanks! > jlc > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/148dec48/attachment.html From ryannyl at gmail.com Tue Dec 1 10:40:26 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 2 Dec 2009 02:40:26 +0800 Subject: [Freeswitch-users] User logon/logout from analog phones Message-ID: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> Dear All: I try to register from a feature code of an analog phone like Elastix. It is useful for DID. There is an idea that I use dynamic dialplan to implement it and it's not really register to FS. And I need to run script to insert or delete dialplan to the database when dialed.(Input logon's ExtNumber and Password) Is that right? or any recommendation? Thanks in advance. -- Sincerely regards, Ryanny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/51a1a06b/attachment.html From msc at freeswitch.org Tue Dec 1 11:04:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 11:04:43 -0800 Subject: [Freeswitch-users] User logon/logout from analog phones In-Reply-To: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> References: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> Message-ID: <87f2f3b90912011104xbbcb86fkc08c9372a34232e6@mail.gmail.com> I'm not entirely sure that I understand your question, so I am going to ask a few questions to clarify. Are you looking to have analog telephones receive incoming calls, like in a call center? Is that why the user of the analog phone would need to log in and log out? I would recommend checkout out mod_xml_curl for the dynamic dialplan. -MC On Tue, Dec 1, 2009 at 10:40 AM, Ryanny Lin wrote: > Dear All: > > I try to register from a feature code of an analog phone like Elastix. > It is useful for DID. > There is an idea that I use dynamic dialplan to implement it and it's not > really register to FS. > And I need to run script to insert or delete dialplan to the database when > dialed.(Input logon's ExtNumber and Password) > Is that right? or any recommendation? > > Thanks in advance. > -- > Sincerely regards, > Ryanny > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/62642775/attachment-0001.html From john_platts at hotmail.com Tue Dec 1 11:19:51 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 1 Dec 2009 13:19:51 -0600 Subject: [Freeswitch-users] Problem with compiling revision 15739 Message-ID: I attempted to do a make current with revision 15739, but some of the Sofia source files will not compile with revision 15739. Those source files were not changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile FreeSWITCH. I used the following to get revision 15738, which was the previous revision, built: make update-clean svn update -r 15738 make all install This does the same stuff as make current, except that revision 15738 is checked out of the SVN repository. _________________________________________________________________ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 From moises.silva at gmail.com Tue Dec 1 11:22:05 2009 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 1 Dec 2009 14:22:05 -0500 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> Message-ID: On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote: > > > 2009/12/1 Moises Silva > > On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: >> >>> Dear All, >>> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). >>> I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) >>> and freeswitch 1.0.5 pre 7 >>> >>> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz >>> and 2 GB memmory. >>> when i start freeswitch if enable echo cencel in zt.conf CPU peak to >>> 100% when i disable echo cancel FS can start but when first call incominng >>> got 100% again. >>> >>> >>> Someone can help me? >>> >>> >>> >> I suggest you to start using freeswitch trunk (and openzap trunk). >> >> > I'll try > > >> In the other hand, when you talk about 4E1 R2, you mean R2 as the >> telephony signaling or is this some kind of board brand? >> > > > telephony signaling > Does the signaling stack comes with the board and is integrated into freeswitch as an endpoint? if not, your only chance is using openr2, which requires freeswitch, openzap and openr2 trunk. http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/23d5c8c2/attachment.html From anthony.minessale at gmail.com Tue Dec 1 11:57:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 13:57:36 -0600 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> Message-ID: <191c3a030912011157o1c0e1d3cp37ffe1338373d60b@mail.gmail.com> the updating of the display code is significantly improved in trunk. Please figure out your email problem and use that. Most likely you need an alternate configuration. What mailer client are you using in switch.conf.xml ? On Tue, Dec 1, 2009 at 11:11 AM, Yehavi Bourvine wrote: > It is MODAPP-373. > > Thanks, __yehavi: > > 2009/12/1 Michael Jerris > > What is the jira bug number on this voicemail email issue? I don't >> recall seeing it. >> >> Mike >> >> On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine >> wrote: >> >> > > Are you on SVN trunk? As far as I recall the callee_id_number/name >> > stuff isnt in 1.0.4. >> > >> > No, because the SVN has problems with Emailing the voicemail... >> > >> > We use 1.0.4 and set sip_callee_id_number/name which works. I would >> > like to not set it and get it from the other side... >> > >> > Thanks! __Yehavi: >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/f48b6771/attachment.html From anthony.minessale at gmail.com Tue Dec 1 11:59:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 13:59:09 -0600 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> Message-ID: <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> yes he did you can see it in his trace. you can not use both of them together...... On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > The only way this would happen would be if this is set to proxy media not > bypass. Are you setting both? > > Mike > > On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > > In the following trace, 102 is FS IP, 104 is calling party and 13 is > called party. > > with bypass_media, FS still changes c=IN IP4 192.168.1.102 > > Any idea why? > > > freeswitch at localhost.localdomain> recv 951 bytes from > udp/[192.168.1.104]:5060 at 22:56:33.782715: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.102 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport > From: > >;tag=786224322 > To: > > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > Contact: > Content-Type: application/sdp > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > Max-Forwards: 70 > Supported: 100rel, replaces > User-Agent: SIPPER for PhonerLite > Content-Length: 397 > > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendrecv > ------------------------------------------------------------------------ > send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 > From: > >;tag=786224322 > To: > > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ > PhonerLite at 192.168.1.102 entering state [received][100] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > > EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() > 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259708193783162] > Caller-Channel-Created-Time: [1259708193783162] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [786224322] > variable_sofia_profile_name: [internal] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] > variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_bypass_media: [true] > variable_proxy_media: [true] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_endpoint_disposition: [RECEIVED_NOMEDIA] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > > > 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ > 90964111 at 192.168.1.116:9390 Patched SDP > --- > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > +++ > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 > (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ > 90964111 at 192.168.1.116:9390 SOFIA ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to > sleep > send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g > Max-Forwards: 69 > From: "PhonerLite" > >;tag=8tH6Xjt2XaU9F > To: > Call-ID: 9d052856-596f-122d-1b98-0022190e9476 > CSeq: 123735760 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 404 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/d2f9cc74/attachment-0001.html From mike at jerris.com Tue Dec 1 12:02:17 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 15:02:17 -0500 Subject: [Freeswitch-users] Problem with compiling revision 15739 In-Reply-To: References: Message-ID: I think I just fixed this a few minutes ago, it is running test builds on the build servers now to verify. On Dec 1, 2009, at 2:19 PM, John Platts wrote: > > I attempted to do a make current with revision 15739, but some of the Sofia source files will not compile with revision 15739. Those source files were not changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile FreeSWITCH. I used the following to get revision 15738, which was the previous revision, built: > make update-clean > svn update -r 15738 > make all install > > This does the same stuff as make current, except that revision 15738 is checked out of the SVN repository. > > _________________________________________________________________ > Windows 7: Unclutter your desktop. Learn more. > http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ryannyl at gmail.com Tue Dec 1 12:27:10 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 2 Dec 2009 04:27:10 +0800 Subject: [Freeswitch-users] User logon/logout from analog phones Message-ID: <4bfcac7e0912011227y58204c84ld0611d50afe6237a@mail.gmail.com> Dear Michael: Yes, I want to distribute a real phone number to each analog phone (direct inward dialing). One FXS one analog phone. I guess the user maybe want to add a number mapping this FXS port. Thank you, Michael. mod_xml_curl is really a powerful module. :D ---------- Forward ---------- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Date: Tue, 1 Dec 2009 11:04:43 -0800 I'm not entirely sure that I understand your question, so I am going to ask a few questions to clarify. Are you looking to have analog telephones receive incoming calls, like in a call center? Is that why the user of the analog phone would need to log in and log out? I would recommend checkout out mod_xml_curl for the dynamic dialplan. -MC On Tue, Dec 1, 2009 at 10:40 AM, Ryanny Lin wrote: > Dear All: > > I try to register from a feature code of an analog phone like Elastix. > It is useful for DID. > There is an idea that I use dynamic dialplan to implement it and it's not > really register to FS. > And I need to run script to insert or delete dialplan to the database when > dialed.(Input logon's ExtNumber and Password) > Is that right? or any recommendation? > > Thanks in advance. > -- > Sincerely regards, > Ryanny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/eededd36/attachment.html From john_platts at hotmail.com Tue Dec 1 12:46:02 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 1 Dec 2009 14:46:02 -0600 Subject: [Freeswitch-users] Blind transfer fails in FreeSWITCH, even if proxying and media bypass are enabled Message-ID: I have tried to do a blind transfer from a phone that is registered with FreeSWITCH, and it will fail, even when proxying and media bypass are enabled. Details about this issue can be found here: http://jira.freeswitch.org/browse/MODENDP-272 _________________________________________________________________ Get gifts for them and cashback for you. Try Bing now. http://www.bing.com/shopping/search?q=xbox+games&scope=cashback&form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_Shopping_Giftsforthem_cashback_1x1 From dome at tel.co.th Tue Dec 1 12:47:36 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Dec 2009 03:47:36 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> Message-ID: <8ccbff060912011247j318ca089n6f0c9fa39c417dcd@mail.gmail.com> 2009/12/2 Moises Silva > > > On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote: > >> >> >> 2009/12/1 Moises Silva >> >> On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: >>> >>>> Dear All, >>>> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ >>>> card). I use zaptel driver from http://e400p.phoniceq.com/driver/(1.4.12.1) and freeswitch 1.0.5 pre 7 >>>> >>>> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz >>>> and 2 GB memmory. >>>> when i start freeswitch if enable echo cencel in zt.conf CPU peak to >>>> 100% when i disable echo cancel FS can start but when first call incominng >>>> got 100% again. >>>> >>>> >>>> Someone can help me? >>>> >>>> >>>> >>> I suggest you to start using freeswitch trunk (and openzap trunk). >>> >>> >> I'll try >> >> >>> In the other hand, when you talk about 4E1 R2, you mean R2 as the >>> telephony signaling or is this some kind of board brand? >>> >> >> >> telephony signaling >> > > Does the signaling stack comes with the board and is integrated into > freeswitch as an endpoint? if not, your only chance is using openr2, which > requires freeswitch, openzap and openr2 trunk. > > http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 > yes i use openr2. I'm setup follow wiki. Dome C. > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0e241d08/attachment.html From b0ef at esben-stien.name Tue Dec 1 14:27:49 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Tue, 01 Dec 2009 23:27:49 +0100 Subject: [Freeswitch-users] Freeswitch Video Capture and Playback In-Reply-To: <87k4xlga1k.fsf@quasar.esben-stien.name> (Esben Stien's message of "Fri\, 20 Nov 2009 04\:16\:55 +0100") References: <87k4xlga1k.fsf@quasar.esben-stien.name> Message-ID: <87pr6yz5wa.fsf@quasar.esben-stien.name> Esben Stien writes: > trying to record and play back video So nobody is using video with freeswitch?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From b0ef at esben-stien.name Tue Dec 1 14:30:09 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Tue, 01 Dec 2009 23:30:09 +0100 Subject: [Freeswitch-users] Ring Forever Message-ID: <87ljhmz5se.fsf@quasar.esben-stien.name> I'd like to set up an extension that would just ring forever. When a person calls this extension, it would ring until the end of times. I've tried several ways to do this, without luck, and I don't find any information on this on the wiki. Any pointers how to do this?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From erandr-junk at usa.net Tue Dec 1 13:51:42 2009 From: erandr-junk at usa.net (eaf) Date: Tue, 1 Dec 2009 13:51:42 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <26594250.post@talk.nabble.com> Hi, I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how it can be programmed), but ran into one issue with sound quality that I just cannot workaround by myself. I would describe the sound problem as being "choppy". From time to time small portions of the other party's voice are dropped, so the voice kind of stutters. This is not too bad, but is really noticeable, happens in every call and I don't experience the same with Asterisk running on the same box. I attached two files: freeswitch.wav and asterisk.mp3 to illustrate my point. Issue completely goes away, if I set inbound-proxy-media to true. The way how I test is to connect SPA-2000 via 10mbps LAN to the box directly exposed to internet, and then dial a toll-free via FutureNine (a SIP provider). The codec in use is PCMU. Can't really try PCMA or anything else with this provider. Only PCMU. Tried to match ptime of provider (30) with ptime of the SPA, didn't get any improvement. Tried turning off recording, no change either. What puzzles me is that even with greedy codec negotiations and with PCMU on both sides of FreeSWITCH, it's still saying that TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to illustrate. The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that it's not a performance issue. http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log Tried both 1.0.4 and 1.0.5pre5. Same results. What should I do next? Calls are consistently bad with FreeSWITCH, and consistently show no glitches with Asterisk. -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26594250.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From erandr-junk at usa.net Tue Dec 1 13:52:03 2009 From: erandr-junk at usa.net (eaf) Date: Tue, 1 Dec 2009 13:52:03 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <26599565.post@talk.nabble.com> I should also add, after browsing through some topics here, that my SIP provider sends 172-byte RTP frames, which is in accordance with ptime:20 that it gives to FreeSWITCH. eaf wrote: > > Hi, > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how > it can be programmed), but ran into one issue with sound quality that I > just cannot workaround by myself. I would describe the sound problem as > being "choppy". From time to time small portions of the other party's > voice are dropped, so the voice kind of stutters. This is not too bad, but > is really noticeable, happens in every call and I don't experience the > same with Asterisk running on the same box. I attached two files: > freeswitch.wav and asterisk.mp3 to illustrate my point. > > Issue completely goes away, if I set inbound-proxy-media to true. > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > directly exposed to internet, and then dial a toll-free via FutureNine (a > SIP provider). > > The codec in use is PCMU. Can't really try PCMA or anything else with this > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > the SPA, didn't get any improvement. Tried turning off recording, no > change either. > > What puzzles me is that even with greedy codec negotiations and with PCMU > on both sides of FreeSWITCH, it's still saying that > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to > illustrate. > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > it's not a performance issue. > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > What should I do next? Calls are consistently bad with FreeSWITCH, and > consistently show no glitches with Asterisk. > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Tue Dec 1 14:05:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 16:05:45 -0600 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <87ljhmz5se.fsf@quasar.esben-stien.name> References: <87ljhmz5se.fsf@quasar.esben-stien.name> Message-ID: <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> do you want to generate ringback forever or no? The calling party will probably abort at some point. put both of these in your context then use one of these 2 sets of actions in your main ext On Tue, Dec 1, 2009 at 4:30 PM, Esben Stien wrote: > I'd like to set up an extension that would just ring forever. > > When a person calls this extension, it would ring until the end of > times. > > I've tried several ways to do this, without luck, and I don't find any > information on this on the wiki. > > Any pointers how to do this?. > > -- > Esben Stien is b0ef at e s a > http://www. s t n m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@ n n > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/3d01646d/attachment-0001.html From JCasale at activenetwerx.com Tue Dec 1 14:09:43 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 1 Dec 2009 22:09:43 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> Message-ID: >In this case, the $1 will only contain whatever is in the parens in your expression, i.e. > > >What do you have for your expression? >-MC Well, untested of course as I am busy with school:) But what I wrote up to try at Christmas (with your addition) was: Am I correct in presuming that Freeswitch will answer a fax from a local zap based user just like it does from an FXO port connected to a POTS line? What I hope to do here is catch any call made from that extension (the zap based fax machine/user) and push its call into the fax module. Thanks for taking the time to help! jlc From msc at freeswitch.org Tue Dec 1 14:22:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 14:22:53 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> Message-ID: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale wrote: > >In this case, the $1 will only contain whatever is in the parens in your > expression, i.e. > > > > > >What do you have for your expression? > >-MC > > Well, untested of course as I am busy with school:) But what I wrote up to > try at > Christmas (with your addition) was: > > > > > > > > data="/opt/freeswitch/scripts/emailfax.sh $1 at fax.com/tmp/${uuid}.rxfax.tiff"/> > > > > > Am I correct in presuming that Freeswitch will answer a fax from a local > zap based user > just like it does from an FXO port connected to a POTS line? What I hope to > do here is > catch any call made from that extension (the zap based fax machine/user) > and push its > call into the fax module. > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that dials. This extension looks ok. Try it out and let us know how it goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/927084aa/attachment.html From anthony.minessale at gmail.com Tue Dec 1 14:29:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 16:29:26 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26599565.post@talk.nabble.com> References: <26599565.post@talk.nabble.com> Message-ID: <191c3a030912011429p2cc7d834t9ca2b56b45682fe6@mail.gmail.com> linksys has had a bug for eons that can be fixed by setting the ptime (or rtp packet size in their terms) in it's firmware to .20 instead of .30 Asterisk does not use async RTP like we do so it's never a problem you can disable the timer by setting the channel var rtp_timer_name=none or sofia param rtp-timer-name to none in the sofia profile. You should also test this on latest SVN trunk or wait for pre8 On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > I should also add, after browsing through some topics here, that my SIP > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > that it gives to FreeSWITCH. > > > eaf wrote: > > > > Hi, > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > how > > it can be programmed), but ran into one issue with sound quality that I > > just cannot workaround by myself. I would describe the sound problem as > > being "choppy". From time to time small portions of the other party's > > voice are dropped, so the voice kind of stutters. This is not too bad, > but > > is really noticeable, happens in every call and I don't experience the > > same with Asterisk running on the same box. I attached two files: > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > directly exposed to internet, and then dial a toll-free via FutureNine (a > > SIP provider). > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > this > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > > the SPA, didn't get any improvement. Tried turning off recording, no > > change either. > > > > What puzzles me is that even with greedy codec negotiations and with PCMU > > on both sides of FreeSWITCH, it's still saying that > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > to > > illustrate. > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > it's not a performance issue. > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > consistently show no glitches with Asterisk. > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/09f705a3/attachment.html From peter at cindyandpeter.com Tue Dec 1 14:41:38 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 17:41:38 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: <00ba01ca72d7$73272200$59756600$@com> Just remove the terminating '/' at the end of the second condition tag.... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 01, 2009 5:23 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Faxing Advice On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale wrote: >In this case, the $1 will only contain whatever is in the parens in your expression, i.e. > > >What do you have for your expression? >-MC Well, untested of course as I am busy with school:) But what I wrote up to try at Christmas (with your addition) was: Am I correct in presuming that Freeswitch will answer a fax from a local zap based user just like it does from an FXO port connected to a POTS line? What I hope to do here is catch any call made from that extension (the zap based fax machine/user) and push its call into the fax module. Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that dials. This extension looks ok. Try it out and let us know how it goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1131e3e6/attachment-0001.html From JCasale at activenetwerx.com Tue Dec 1 15:08:06 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 1 Dec 2009 23:08:06 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <00ba01ca72d7$73272200$59756600$@com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> <00ba01ca72d7$73272200$59756600$@com> Message-ID: >Just remove the terminating '/' at the end of the second condition tag.... > > I tried to see based on examples if it was obvious to me why that should not be there but it didn't jump out:) Cuold you explain that please? The gateway will arrive at the end of the week, but I probably won't get to this now until Christmas as I missed my opportunity:) Once I get it working, I will update the wiki as I am sure it's useful to many others. Thanks everyone! jlc From Russell.Mosemann at cune.org Tue Dec 1 15:21:33 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 1 Dec 2009 23:21:33 -0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: Message-ID: <20091201232133.76DAB4309FA@mail.cune.org> "Joseph L. Casale" said: > >Just remove the terminating '/' at the end of the second condition tag.... > > > > > > I tried to see based on examples if it was obvious to me why that should not be there > but it didn't jump out:) Cuold you explain that please? It is a multi-line condition. If a condition is only one line long, it begins and ends on the same line. The "/>" combination is the sequence that closes the tag. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From b0ef at esben-stien.name Tue Dec 1 16:29:25 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Wed, 02 Dec 2009 01:29:25 +0100 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> (Anthony Minessale's message of "Tue\, 1 Dec 2009 16\:05\:45 -0600") References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> Message-ID: <87einexlp6.fsf@quasar.esben-stien.name> Anthony Minessale writes: > do you want to generate ringback forever or no? Yes, I want the dialing party to get a ring forever. This is because I cannot transfer the party to my SIP phone, because my SIP phone is broken for incoming calls. I'll solve it by letting the party hear a ring and then I'll use the event socket to give me a notification. I'll then dial in with my SIP phone to a conference room and transfer the calling party there. I did this: ..which gives me an eternal ring, but it sounds choppy and it sounds as if the following ring starts on top of the currently playing ring. I hear crackling sounds, which I don't experience with any other use of freeswitch. Any idea? -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From anthony.minessale at gmail.com Tue Dec 1 15:39:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 17:39:14 -0600 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <87einexlp6.fsf@quasar.esben-stien.name> References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> <87einexlp6.fsf@quasar.esben-stien.name> Message-ID: <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> are you on older REV? try answering first to compare. On Tue, Dec 1, 2009 at 6:29 PM, Esben Stien wrote: > Anthony Minessale writes: > > > do you want to generate ringback forever or no? > > Yes, I want the dialing party to get a ring forever. This is because I > cannot transfer the party to my SIP phone, because my SIP phone is > broken for incoming calls. > > I'll solve it by letting the party hear a ring and then I'll use the > event socket to give me a notification. I'll then dial in with my SIP > phone to a conference room and transfer the calling party there. > > I did this: > > > > > > > > > > > > > > > > ..which gives me an eternal ring, but it sounds choppy and it sounds as > if the following ring starts on top of the currently playing ring. > > I hear crackling sounds, which I don't experience with any other use of > freeswitch. > > Any idea? > > -- > Esben Stien is b0ef at e s a > http://www. s t n m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@ n n > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/ce79f411/attachment.html From b0ef at esben-stien.name Tue Dec 1 17:24:11 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Wed, 02 Dec 2009 02:24:11 +0100 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> (Anthony Minessale's message of "Tue\, 1 Dec 2009 17\:39\:14 -0600") References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> <87einexlp6.fsf@quasar.esben-stien.name> <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> Message-ID: <87einew4lg.fsf@quasar.esben-stien.name> Anthony Minessale writes: > are you on older REV? I think I'm on 15334 > try answering first to compare. ?, that's what I do: -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From msc at freeswitch.org Tue Dec 1 16:56:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 16:56:25 -0800 Subject: [Freeswitch-users] FreeSWITCH Survey: What Environment Do You Normally Use? Message-ID: <87f2f3b90912011656m5d08d466n641e341ec9889373@mail.gmail.com> Hi folks, I'm doing a little survey to get an idea of what everyone prefers to use for their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc. Please log in to the main page and check out this node: http://www.freeswitch.org/node/206 Select the environment that you use the most or prefer to use. In a week or so I will send out the final tally. I'm sure the only question is who will come in second place after 64-bit CentOS/Red Hat. :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1d71d6b5/attachment.html From anthony.minessale at gmail.com Tue Dec 1 17:02:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 19:02:11 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: References: Message-ID: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: > Sure, I'll try that. I'm just building freeswitch-snapshot that I > downloaded from files.freeswitch.org > > I also experience, when bridging a call from an FXS to FXO the call is cut > after a random time (this does not appear when bridging SIP to FXO). Might > this upgrade fix this problem also ? > > > > Fran?ois > > > > On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: > > can you test svn trunk or latest pre release of 1.0.5 > > > On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: > >> Hello, >> >> >> >> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP >> problems on the FXS ports. >> >> When I ring on FXS ports, the connected phone does not display >> callerid/callerid-name. >> >> I tried turning the stuff of in openzap.conf.xml () but it did not help. >> >> >> >> As a side note, turning this on on the FXO ports drops the callerid >> information on incoming calls. >> >> >> >> Running freeswitch 1.0.4 on linux 2.6.27. >> >> >> >> Fran?ois >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/edb26ac0/attachment-0001.html From erandr-junk at usa.net Tue Dec 1 17:26:58 2009 From: erandr-junk at usa.net (erandr-junk at usa.net) Date: Tue, 01 Dec 2009 20:26:58 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <927NLBBZ73918S04.1259717218@cmsweb04.cms.usa.net> Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just installing SVN trunk didn't fix it either, but setting that option afterwards surely did the trick. One thing I've noticed while staring at the console is that it *looks like* that w/o the new setting the stuttering happens when FS either re-registers itself with the provider or one of the SPA's port re-registers with FS. ------ Original Message ------ Received: Tue, 01 Dec 2009 05:33:26 PM EST From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Choppy sound with PCMU > linksys has had a bug for eons that can be fixed by setting the ptime (or > rtp packet size in their terms) > in it's firmware to .20 instead of .30 > > Asterisk does not use async RTP like we do so it's never a problem > you can disable the timer by setting the channel var rtp_timer_name=none or > sofia param rtp-timer-name to none in the sofia profile. > > You should also test this on latest SVN trunk or wait for pre8 > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > I should also add, after browsing through some topics here, that my SIP > > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > > that it gives to FreeSWITCH. > > > > > > eaf wrote: > > > > > > Hi, > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > > how > > > it can be programmed), but ran into one issue with sound quality that I > > > just cannot workaround by myself. I would describe the sound problem as > > > being "choppy". From time to time small portions of the other party's > > > voice are dropped, so the voice kind of stutters. This is not too bad, > > but > > > is really noticeable, happens in every call and I don't experience the > > > same with Asterisk running on the same box. I attached two files: > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > directly exposed to internet, and then dial a toll-free via FutureNine (a > > > SIP provider). > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > > this > > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > > > the SPA, didn't get any improvement. Tried turning off recording, no > > > change either. > > > > > > What puzzles me is that even with greedy codec negotiations and with PCMU > > > on both sides of FreeSWITCH, it's still saying that > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > > to > > > illustrate. > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > > it's not a performance issue. > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > > consistently show no glitches with Asterisk. > > > > > > > > > > -- > > View this message in context: > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peter at cindyandpeter.com Tue Dec 1 18:26:26 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 21:26:26 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <20091201232133.76DAB4309FA@mail.cune.org> References: <20091201232133.76DAB4309FA@mail.cune.org> Message-ID: <00dd01ca72f6$da19ddd0$8e4d9970$@com> To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. ... content ... If there is no content, you get: Or, on one line, . You're allowed to abbreviate that to just . So in your case: <--- these are themselves abbreviations of !! Hope that helps! Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org Sent: Tuesday, December 01, 2009 6:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Faxing Advice "Joseph L. Casale" said: > >Just remove the terminating '/' at the end of the second condition tag.... > > > > > > I tried to see based on examples if it was obvious to me why that should not be there > but it didn't jump out:) Cuold you explain that please? It is a multi-line condition. If a condition is only one line long, it begins and ends on the same line. The "/>" combination is the sequence that closes the tag. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From JCasale at activenetwerx.com Tue Dec 1 19:02:13 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 2 Dec 2009 03:02:13 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <00dd01ca72f6$da19ddd0$8e4d9970$@com> References: <20091201232133.76DAB4309FA@mail.cune.org> <00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: >To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. > > ... content ... > /snip Ahh, so must all the actions be contained within at least one condition tag as content, or could have I kept the last "/" on the last condition and dropped the line? Thanks everyone :) jlc From juanbackson at gmail.com Tue Dec 1 19:11:37 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Dec 2009 11:11:37 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> Message-ID: <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Hi, I also did try to set only bypass_media, but it still does not work? freeswitch still modifies the c= line, causing the call to fail. Could someone please help? send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a Max-Forwards: 69 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/ 90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [calling][0] 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/ 90964111 at 192.168.1.13:5060 SOFIA ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [proceeding][180] 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/external_1/90964111 at 192.168.1.13:5060! 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/external_1/PhonerLite at 192.168.1.102! 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 From: >;tag=2454193703 To: >;tag=FFKXgjN02m02N Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [early][180] 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 256 v=0 o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10096 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp Max-Forwards: 70 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [ready][200] 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10096 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/ 90964111 at 192.168.1.13:5060] has been answered 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/ 90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 From: >;tag=2454193703 To: >;tag=FFKXgjN02m02N2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 originate_disposition=[SUCCESS] Content-Type: application/sdp Content-Disposition: session 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] Content-Length: 207 v=0 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] m=audio 10096 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/ 90964111 at 192.168.1.13:5060 SOFIA HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next route... EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_HIBERNATE] Channel-State-Number: [8] Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [answered] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259751417450336] Caller-Channel-Created-Time: [1259751417450336] Caller-Channel-Answered-Time: [1259751419381828] Caller-Channel-Progress-Time: [1259751417736234] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2454193703] variable_sofia_profile_name: [external_1] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 794697697 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ >;tag=2454193703 To: >;tag=FFKXgjN02m02N Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] Contact: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/ 90964111 at 192.168.1.116:9392) 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/ PhonerLite at 192.168.1.102 receive message [MEDIA] 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/ PhonerLite at 192.168.1.102 reading on a session with no media! 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [completed][200] 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [ready][200] 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/ 90964111 at 192.168.1.13:5060 receive message [MEDIA] 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj Max-Forwards: 69 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757373 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 30632 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [calling][0] send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: ------------------------------------------------------------------------ On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > yes he did you can see it in his trace. > you can not use both of them together...... > > > > On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > >> The only way this would happen would be if this is set to proxy media not >> bypass. Are you setting both? >> >> Mike >> >> On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: >> >> In the following trace, 102 is FS IP, 104 is calling party and 13 is >> called party. >> >> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >> >> Any idea why? >> >> >> freeswitch at localhost.localdomain> recv 951 bytes from >> udp/[192.168.1.104]:5060 at 22:56:33.782715: >> >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.104:5060 >> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >> From: >> >;tag=786224322 >> To: > >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> Contact: >> Content-Type: application/sdp >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> Max-Forwards: 70 >> Supported: 100rel, replaces >> User-Agent: SIPPER for PhonerLite >> Content-Length: 397 >> >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.104:5060 >> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >> From: >> >;tag=786224322 >> To: > >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel >> sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ >> PhonerLite at 192.168.1.102 entering state [received][100] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> >> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_EXECUTE] >> Channel-State-Number: [4] >> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [ringing] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259708193783162] >> Caller-Channel-Created-Time: [1259708193783162] >> Caller-Channel-Answered-Time: [0] >> Caller-Channel-Progress-Time: [0] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [786224322] >> variable_sofia_profile_name: [internal] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> ] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_bypass_media: [true] >> variable_proxy_media: [true] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> >> >> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ >> 90964111 at 192.168.1.116:9390 Patched SDP >> --- >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> +++ >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >> (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to >> sleep >> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >> >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >> Max-Forwards: 69 >> From: "PhonerLite" >> >;tag=8tH6Xjt2XaU9F >> To: >> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >> CSeq: 123735760 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 404 >> Remote-Party-ID: "PhonerLite" >> >;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/963a7360/attachment-0001.html From erandr-junk at usa.net Tue Dec 1 19:19:39 2009 From: erandr-junk at usa.net (erandr-junk at usa.net) Date: Tue, 01 Dec 2009 22:19:39 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> Wow... Thinking about this timer setting and about how it converted send()/recv() from non-blocking to blocking, I straced freeswitch when it was supposed to be idle. It never pauses! It keeps going in and out of select() every millisecond! Why?? ------ Original Message ------ Received: Tue, 01 Dec 2009 08:31:46 PM EST From: erandr-junk at usa.net To: Subject: Re: [Freeswitch-users] Choppy sound with PCMU > Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just > installing SVN trunk didn't fix it either, but setting that option afterwards > surely did the trick. > > One thing I've noticed while staring at the console is that it *looks like* > that w/o the new setting the stuttering happens when FS either re-registers > itself with the provider or one of the SPA's port re-registers with FS. > > ------ Original Message ------ > Received: Tue, 01 Dec 2009 05:33:26 PM EST > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > linksys has had a bug for eons that can be fixed by setting the ptime (or > > rtp packet size in their terms) > > in it's firmware to .20 instead of .30 > > > > Asterisk does not use async RTP like we do so it's never a problem > > you can disable the timer by setting the channel var rtp_timer_name=none or > > sofia param rtp-timer-name to none in the sofia profile. > > > > You should also test this on latest SVN trunk or wait for pre8 > > > > > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > > > > I should also add, after browsing through some topics here, that my SIP > > > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > > > that it gives to FreeSWITCH. > > > > > > > > > eaf wrote: > > > > > > > > Hi, > > > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > > > how > > > > it can be programmed), but ran into one issue with sound quality that I > > > > just cannot workaround by myself. I would describe the sound problem as > > > > being "choppy". From time to time small portions of the other party's > > > > voice are dropped, so the voice kind of stutters. This is not too bad, > > > but > > > > is really noticeable, happens in every call and I don't experience the > > > > same with Asterisk running on the same box. I attached two files: > > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > > directly exposed to internet, and then dial a toll-free via FutureNine > (a > > > > SIP provider). > > > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > > > this > > > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime > of > > > > the SPA, didn't get any improvement. Tried turning off recording, no > > > > change either. > > > > > > > > What puzzles me is that even with greedy codec negotiations and with > PCMU > > > > on both sides of FreeSWITCH, it's still saying that > > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > > > to > > > > illustrate. > > > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode > LX800 > > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > > > it's not a performance issue. > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > > > consistently show no glitches with Asterisk. > > > > > > > > > > > > > > -- > > > View this message in context: > > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Dec 1 19:26:27 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 22:26:27 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Message-ID: how are you sending both invites here? can you explain the full call path and how you are originating these calls? On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > Hi, > > I also did try to set?only bypass_media, but it still does not work??freeswitch still modifies the c= line, causing the call to fail. > > Could someone please help? > > > send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > Max-Forwards: 69 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 402 > Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/90964111 at 192.168.1.13:5060 SOFIA ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep > recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [proceeding][180] > 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/external_1/90964111 at 192.168.1.13:5060! > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/external_1/PhonerLite at 192.168.1.102! > 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [early][180] > 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! > recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Supported: replaces > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: > ------------------------------------------------------------------------ > ACK sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp > Max-Forwards: 70 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [ready][200] > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/90964111 at 192.168.1.13:5060] has been answered > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) > EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) > > 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] > send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE > > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 originate_disposition=[SUCCESS] > Content-Type: application/sdp > Content-Disposition: session > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > Content-Length: 207 > > v=0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > m=audio 10096 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() > 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next route... > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() > 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_HIBERNATE] > Channel-State-Number: [8] > Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [answered] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259751417450336] > Caller-Channel-Created-Time: [1259751417450336] > Caller-Channel-Answered-Time: [1259751419381828] > Caller-Channel-Progress-Time: [1259751417736234] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > Other-Leg-Username: [PhonerLite] > Other-Leg-Caller-ID-Name: [PhonerLite] > Other-Leg-Caller-ID-Number: [PhonerLite] > Other-Leg-Network-Addr: [192.168.1.104] > Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] > Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] > Other-Leg-Source: [mod_sofia] > Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] > Other-Leg-Screen-Bit: [true] > Other-Leg-Privacy-Hide-Name: [false] > Other-Leg-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [2454193703] > variable_sofia_profile_name: [external_1] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] > variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 794697697 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > variable_proto_specific_hangup_cause: [sip:403] > variable_sip_hangup_phrase: [Because] > variable_bypass_media: [true] > variable_success_bridge: [true] > variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] > variable_switch_m_sdp: [v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_endpoint_disposition: [ANSWER] > variable_originate_disposition: [SUCCESS] > variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] > variable_current_application: [info] > > recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: > ------------------------------------------------------------------------ > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] > > Contact: > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/90964111 at 192.168.1.116:9392) > 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/PhonerLite at 192.168.1.102 receive message [MEDIA] > 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/PhonerLite at 192.168.1.102 reading on a session with no media! > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [completed][200] > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [ready][200] > 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/90964111 at 192.168.1.13:5060 receive message [MEDIA] > 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: > ------------------------------------------------------------------------ > INVITE sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj > Max-Forwards: 69 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757373 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 223 > Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 30632 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: > ------------------------------------------------------------------------ > > On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale wrote: > yes he did you can see it in his trace. > you can not use both of them together...... > > > > On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > The only way this would happen would be if this is set to proxy media not bypass. Are you setting both? > > Mike > > On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > >> In the following trace, 102 is FS IP, 104 is calling party and 13 is called party. >> >> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >> >> Any idea why? >> >> >> freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 22:56:33.782715: >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >> From: ;tag=786224322 >> To: >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> Contact: >> Content-Type: application/sdp >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> Max-Forwards: 70 >> Supported: 100rel, replaces >> User-Agent: SIPPER for PhonerLite >> Content-Length: 397 >> >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrecv >> ------------------------------------------------------------------------ >> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >> From: ;tag=786224322 >> To: >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [received][100] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> >> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_EXECUTE] >> Channel-State-Number: [4] >> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [ringing] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259708193783162] >> Caller-Channel-Created-Time: [1259708193783162] >> Caller-Channel-Answered-Time: [0] >> Caller-Channel-Progress-Time: [0] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [786224322] >> variable_sofia_profile_name: [internal] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_bypass_media: [true] >> variable_proxy_media: [true] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> >> >> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/90964111 at 192.168.1.116:9390 Patched SDP >> --- >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> +++ >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep >> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >> Max-Forwards: 69 >> From: "PhonerLite" ;tag=8tH6Xjt2XaU9F >> To: >> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >> CSeq: 123735760 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 404 >> Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/a6b9d30c/attachment-0001.html From Russell.Mosemann at cune.org Tue Dec 1 19:34:03 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Tue, 1 Dec 2009 21:34:03 -0600 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <20091201232133.76DAB4309FA@mail.cune.org><00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: Joseph L. Casale wrote: > Ahh, so must all the actions be contained within at least one condition > tag as content, Yes. > or could have I kept the > last "/" on the last condition and dropped the line? No. Think of the tags as a begin/end pair that surround the content. If there is no content, then you can use a one-line condition tag. or stuff -- Russell Mosemann From peter at cindyandpeter.com Tue Dec 1 19:41:32 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 22:41:32 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <20091201232133.76DAB4309FA@mail.cune.org> <00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: <00e401ca7301$5723b580$056b2080$@com> Indeed, all actions must be contained with a condition tag. FS just processes the conditions inside the extension top-to-bottom, so: if the top condition (without actions in it) doesn't match, it stops processing that extension. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph L. Casale Sent: Tuesday, December 01, 2009 10:02 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: [Freeswitch-users] Faxing Advice >To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. > > ... content ... > /snip Ahh, so must all the actions be contained within at least one condition tag as content, or could have I kept the last "/" on the last condition and dropped the line? Thanks everyone :) jlc _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From juanbackson at gmail.com Tue Dec 1 20:07:22 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Dec 2009 12:07:22 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Message-ID: <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> Hi Mike, Here is a very strange SIP outgoing INVITES from freeswitch: The call path is this: 192.168.1.104 (phonerlite) -> 192.168.1.102 ( freeswitch ) 192.168.1.102 -> 192.168.1.116 (sipp gives back 403) 192.168.1.102 -> 192.168.1.13 ( phone ) The first INVITE to 192.168.1.13 has the right c= and o= ( both is pointing to 192.168.1.104). But the for some unknown reason, Freeswitch sends INVITES again. But in the INVITE resend, the o = becomes fs's ip. I have no idea. This is only bypass_meida. Attached is the fs log. ngrep -q -p -W byline port 5060 interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip) and ( port 5060 ) U 192.168.1.104:5060 -> 192.168.1.102:5060 INVITE sip:90964111 at 192.168.1.102 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport. From: >;tag=2563216860. To: >. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . Content-Type: application/sdp. Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE. Max-Forwards: 70. Supported: 100rel, replaces. User-Agent: SIPPER for PhonerLite. Content-Length: 396. . v=0. o=- 478760567 0 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=sendrecv. U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9390 INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. Max-Forwards: 69. From: "PhonerLite" >;tag=r0pv05c0848ae. To: . Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 401. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 523740055483911509 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.116:9390 -> 192.168.1.102:5060 SIP/2.0 403 Because. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. From: "PhonerLite" >;tag=r0pv05c0848ae. To: ;tag=6. Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9390 ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. Max-Forwards: 69. From: "PhonerLite" >;tag=r0pv05c0848ae. To: ;tag=6. Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9391 INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. Max-Forwards: 69. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: . Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 402. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 5392773558290384508 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.116:9391 -> 192.168.1.102:5060 SIP/2.0 403 Because. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: ;tag=6. Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9391 ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. Max-Forwards: 69. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: ;tag=6. Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: . Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 402. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522444 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: . Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Supported: replaces. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE. Content-Type: application/sdp. Content-Length: 256. . v=0. o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. s=A conversation. c=IN IP4 192.168.1.13. t=0 0. m=audio 10098 RTP/AVP 8 0 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 192.168.1.102:5060 -> 192.168.1.13:5060 ACK sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ. Max-Forwards: 70. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 207. . v=0. o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. s=A conversation. c=IN IP4 192.168.1.13. t=0 0. m=audio 10098 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 192.168.1.104:5060 -> 192.168.1.102:5060 ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK00b67e2664ddde1180d3001a805656a5;rport. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 ACK. Contact: . Max-Forwards: 70. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 223. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.102. t=0 0. m=audio 33352 RTP/AVP 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 192.168.1.104:43488 -> 192.168.1.102:5060 . . .............. U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 223. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.102. t=0 0. m=audio 33352 RTP/AVP 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 ACK sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 BYE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e. Max-Forwards: 70. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758964 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma. Max-Forwards: 70. From: >;tag=QQX3yavvBvjrj. To: >;tag=2563216860. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 123758964 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". Content-Length: 0. . U 192.168.1.104:5060 -> 192.168.1.102:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma. From: >;tag=QQX3yavvBvjrj. To: >;tag=2563216860. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 123758964 BYE. Contact: . User-Agent: SIPPER for PhonerLite. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . Content-Length: 0. . On Wed, Dec 2, 2009 at 11:26 AM, Michael Jerris wrote: > how are you sending both invites here? can you explain the full call path > and how you are originating these calls? > > > On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > > Hi, > > I also did try to set only bypass_media, but it still does not work? > freeswitch still modifies the c= line, causing the call to fail. > > Could someone please help? > > > send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > Max-Forwards: 69 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 402 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/ > 90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [calling][0] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 > (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 > (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/ > 90964111 at 192.168.1.13:5060 SOFIA ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 > (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 > (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 > (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 > (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to > sleep > recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [proceeding][180] > 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready > sofia/external_1/90964111 at 192.168.1.13:5060! > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ > PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready > sofia/external_1/PhonerLite at 192.168.1.102! > 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: > >;tag=2454193703 > To: > >;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 > sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [early][180] > 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready > sofia/external_1/PhonerLite at 192.168.1.102! > recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Supported: replaces > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, > UPDATE, MESSAGE > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: > ------------------------------------------------------------------------ > ACK sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp > Max-Forwards: 70 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [ready][200] > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/ > 90964111 at 192.168.1.13:5060] has been answered > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/ > 90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 > 256 25 2591585 1) > EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 > 621 201 256 25 2591585 1) > > 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel > [sofia/external_1/PhonerLite at 192.168.1.102] has been answered > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate > Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] > send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: > >;tag=2454193703 > To: >;tag=FFKXgjN02m02N2009-12-02 > 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ > PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE > > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 > originate_disposition=[SUCCESS] > Content-Type: application/sdp > Content-Disposition: session > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > Content-Length: 207 > > v=0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 > (sofia/external_1/90964111 at 192.168.1.13:5060) State Change > CS_CONSUME_MEDIA -> CS_HIBERNATE > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > m=audio 10096 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 > (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/ > 90964111 at 192.168.1.13:5060 SOFIA HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 > sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 > (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to > sleep > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() > 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next > route... > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() > 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_HIBERNATE] > Channel-State-Number: [8] > Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [answered] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259751417450336] > Caller-Channel-Created-Time: [1259751417450336] > Caller-Channel-Answered-Time: [1259751419381828] > Caller-Channel-Progress-Time: [1259751417736234] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > Other-Leg-Username: [PhonerLite] > Other-Leg-Caller-ID-Name: [PhonerLite] > Other-Leg-Caller-ID-Number: [PhonerLite] > Other-Leg-Network-Addr: [192.168.1.104] > Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] > Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] > Other-Leg-Source: [mod_sofia] > Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] > Other-Leg-Screen-Bit: [true] > Other-Leg-Privacy-Hide-Name: [false] > Other-Leg-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [2454193703] > variable_sofia_profile_name: [external_1] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] > variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 794697697 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > variable_proto_specific_hangup_cause: [sip:403] > variable_sip_hangup_phrase: [Because] > variable_bypass_media: [true] > variable_success_bridge: [true] > variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] > variable_switch_m_sdp: [v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_endpoint_disposition: [ANSWER] > variable_originate_disposition: [SUCCESS] > variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] > variable_current_application: [info] > > recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: > ------------------------------------------------------------------------ > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport > From: > >;tag=2454193703 > To: > >;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 > sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] > > Contact: > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/ > 90964111 at 192.168.1.116:9392) > 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/ > PhonerLite at 192.168.1.102 receive message [MEDIA] > 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/ > PhonerLite at 192.168.1.102 reading on a session with no media! > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [completed][200] > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [ready][200] > 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/ > 90964111 at 192.168.1.13:5060 receive message [MEDIA] > 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: > ------------------------------------------------------------------------ > INVITE sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj > Max-Forwards: 69 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757373 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 223 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 30632 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [calling][0] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: > ------------------------------------------------------------------------ > > On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> yes he did you can see it in his trace. >> you can not use both of them together...... >> >> >> >> On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: >> >>> The only way this would happen would be if this is set to proxy media not >>> bypass. Are you setting both? >>> >>> Mike >>> >>> On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: >>> >>> In the following trace, 102 is FS IP, 104 is calling party and 13 is >>> called party. >>> >>> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >>> >>> Any idea why? >>> >>> >>> freeswitch at localhost.localdomain> recv 951 bytes from >>> udp/[192.168.1.104]:5060 at 22:56:33.782715: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.104:5060 >>> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >>> From: >>> >;tag=786224322 >>> To: > >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> Contact: >>> Content-Type: application/sdp >>> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, >>> UPDATE >>> Max-Forwards: 70 >>> Supported: 100rel, replaces >>> User-Agent: SIPPER for PhonerLite >>> Content-Length: 397 >>> >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=sendrecv >>> >>> ------------------------------------------------------------------------ >>> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.1.104:5060 >>> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >>> From: >>> >;tag=786224322 >>> To: > >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel >>> sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ >>> PhonerLite at 192.168.1.102 entering state [received][100] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> >>> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >>> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >>> Channel-State: [CS_EXECUTE] >>> Channel-State-Number: [4] >>> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Call-Direction: [inbound] >>> Presence-Call-Direction: [inbound] >>> Answer-State: [ringing] >>> Caller-Username: [PhonerLite] >>> Caller-Dialplan: [class4] >>> Caller-Caller-ID-Name: [PhonerLite] >>> Caller-Caller-ID-Number: [PhonerLite] >>> Caller-Network-Addr: [192.168.1.104] >>> Caller-Destination-Number: [90964111] >>> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Caller-Source: [mod_sofia] >>> Caller-Context: [default] >>> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Caller-Profile-Index: [1] >>> Caller-Profile-Created-Time: [1259708193783162] >>> Caller-Channel-Created-Time: [1259708193783162] >>> Caller-Channel-Answered-Time: [0] >>> Caller-Channel-Progress-Time: [0] >>> Caller-Channel-Progress-Media-Time: [0] >>> Caller-Channel-Hangup-Time: [0] >>> Caller-Channel-Transfer-Time: [0] >>> Caller-Screen-Bit: [true] >>> Caller-Privacy-Hide-Name: [false] >>> Caller-Privacy-Hide-Number: [false] >>> variable_sip_received_ip: [192.168.1.104] >>> variable_sip_received_port: [5060] >>> variable_sip_via_protocol: [udp] >>> variable_sip_from_user: [PhonerLite] >>> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >>> variable_sip_from_host: [192.168.1.102] >>> variable_sip_from_user_stripped: [PhonerLite] >>> variable_sip_from_tag: [786224322] >>> variable_sofia_profile_name: [internal] >>> variable_sip_req_user: [90964111] >>> variable_sip_req_uri: [90964111 at 192.168.1.102] >>> variable_sip_req_host: [192.168.1.102] >>> variable_sip_to_user: [90964111] >>> variable_sip_to_uri: [90964111 at 192.168.1.102] >>> variable_sip_to_host: [192.168.1.102] >>> variable_sip_contact_port: [5060] >>> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >>> variable_sip_contact_host: [192.168.1.104] >>> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >>> variable_sip_call_id: [ >>> 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] >>> variable_sip_user_agent: [SIPPER for PhonerLite] >>> variable_sip_via_host: [192.168.1.104] >>> variable_sip_via_port: [5060] >>> variable_bypass_media: [true] >>> variable_proxy_media: [true] >>> variable_sip_via_rport: [5060] >>> variable_max_forwards: [70] >>> variable_switch_r_sdp: [v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> ] >>> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >>> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >>> variable_effective_caller_id_number: [PhonerLite] >>> variable_effective_caller_id_name: [PhonerLite] >>> variable_>> variable_routing_digit: [90964111] >>> variable_continue_on_fail: [true] >>> variable_hangup_after_bridge: [true] >>> variable_sip_contact_user: [PhonerLite] >>> >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 Patched SDP >>> --- >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> +++ >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >>> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >>> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >>> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to >>> sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going >>> to sleep >>> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >>> Max-Forwards: 69 >>> From: "PhonerLite" >>> >;tag=8tH6Xjt2XaU9F >>> To: >>> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >>> CSeq: 123735760 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 404 >>> Remote-Party-ID: "PhonerLite" >>> >;party=calling;screen=yes;privacy=off >>> >>> v=0 >>> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/db91a3af/attachment-0001.html -------------- next part -------------- freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 11:49:56.930364: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport From: ;tag=2563216860 To: Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 396 v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 351 bytes to udp/[192.168.1.104]:5060 at 11:49:56.930781: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.930444 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102 [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [received][100] 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3490 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_NEW -> CS_INIT 2009-12-02 19:49:56.930444 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_INIT 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/PhonerLite at 192.168.1.102) State INIT 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:83 sofia/internal/PhonerLite at 192.168.1.102 SOFIA INIT 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:111 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/PhonerLite at 192.168.1.102) State INIT going to sleep 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/PhonerLite at 192.168.1.102) State ROUTING 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:130 sofia/internal/PhonerLite at 192.168.1.102 SOFIA ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:78 sofia/internal/PhonerLite at 192.168.1.102 Standard ROUTING 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_ROUTING -> CS_EXECUTE 2009-12-02 19:49:56.946388 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/PhonerLite at 192.168.1.102) State ROUTING going to sleep 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/PhonerLite at 192.168.1.102) State EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] mod_sofia.c:173 sofia/internal/PhonerLite at 192.168.1.102 SOFIA EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] switch_core_state_machine.c:151 sofia/internal/PhonerLite at 192.168.1.102 Standard EXECUTE EXECUTE sofia/internal/PhonerLite at 192.168.1.102 set(sip_contact_user=PhonerLite) EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 19:49:56.957384 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_INIT 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA INIT 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep send 1156 bytes to udp/[192.168.1.116]:9390 at 11:49:56.960484: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m Max-Forwards: 69 From: "PhonerLite" ;tag=r0pv05c0848ae To: Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 401 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 523740055483911509 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:56.960392 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9390 entering state [calling][0] recv 342 bytes from udp/[192.168.1.116]:9390 at 11:49:56.962288: ------------------------------------------------------------------------ SIP/2.0 403 Because Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m From: "PhonerLite" ;tag=r0pv05c0848ae To: ;tag=6 Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 332 bytes to udp/[192.168.1.116]:9390 at 11:49:56.962463: ------------------------------------------------------------------------ ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m Max-Forwards: 69 From: "PhonerLite" ;tag=r0pv05c0848ae To: ;tag=6 Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.962396 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9390 entering state [terminated][403] 2009-12-02 19:49:56.962396 [NOTICE] sofia.c:3925 Hangup sofia/internal/90964111 at 192.168.1.116:9390 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2009-12-02 19:49:56.962396 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [KILL] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_HANGUP 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9390) State HANGUP 2009-12-02 19:49:56.962396 [DEBUG] mod_sofia.c:306 sofia/internal/90964111 at 192.168.1.116:9390 Overriding SIP cause 603 with 403 from the other leg 2009-12-02 19:49:56.962396 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.116:9390 hanging up, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.116:9390 Standard HANGUP, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9390) State HANGUP going to sleep 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9390) State REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.116:9390 Standard REPORTING, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9390) State REPORTING going to sleep 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:1069 Session 11 (sofia/internal/90964111 at 192.168.1.116:9390) Locked, Waiting on external entities 2009-12-02 19:49:56.963391 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2009-12-02 19:49:56.963391 [INFO] mod_dptools.c:2106 Originate Failed. Cause: CALL_REJECTED 2009-12-02 19:49:56.963391 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: CALL_REJECTED 2009-12-02 19:49:56.963391 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.116:9390] Other-Leg-Unique-ID: [aca42f97-803b-40b4-93b5-79531cdd47e7] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.116:9390] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:56.973380 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [NOTICE] switch_core_session.c:1087 Session 11 (sofia/internal/90964111 at 192.168.1.116:9390) Ended 2009-12-02 19:49:56.974397 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.116:9390 [CS_DESTROY] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9390) State DESTROY 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.116:9390 Standard DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9390) State DESTROY going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_INIT 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9391) State INIT 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA INIT 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9391) State INIT going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9391) State ROUTING 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9391) State ROUTING going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9391) State CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9391) State CONSUME_MEDIA going to sleep send 1157 bytes to udp/[192.168.1.116]:9391 at 11:49:56.980816: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg Max-Forwards: 69 From: "PhonerLite" ;tag=S9FN20X35DZXS To: Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 5392773558290384508 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:56.980394 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9391 entering state [calling][0] recv 342 bytes from udp/[192.168.1.116]:9391 at 11:49:56.990534: ------------------------------------------------------------------------ SIP/2.0 403 Because Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg From: "PhonerLite" ;tag=S9FN20X35DZXS To: ;tag=6 Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 332 bytes to udp/[192.168.1.116]:9391 at 11:49:56.990729: ------------------------------------------------------------------------ ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg Max-Forwards: 69 From: "PhonerLite" ;tag=S9FN20X35DZXS To: ;tag=6 Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.990410 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9391 entering state [terminated][403] 2009-12-02 19:49:56.990410 [NOTICE] sofia.c:3925 Hangup sofia/internal/90964111 at 192.168.1.116:9391 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2009-12-02 19:49:56.990410 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [KILL] 2009-12-02 19:49:56.990410 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.990410 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:49:56.990410 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_HANGUP 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9391) State HANGUP 2009-12-02 19:49:56.991392 [DEBUG] mod_sofia.c:306 sofia/internal/90964111 at 192.168.1.116:9391 Overriding SIP cause 603 with 403 from the other leg 2009-12-02 19:49:56.991392 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.116:9391 hanging up, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.116:9391 Standard HANGUP, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9391) State HANGUP going to sleep 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9391) State REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.116:9391 Standard REPORTING, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9391) State REPORTING going to sleep 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:1069 Session 12 (sofia/internal/90964111 at 192.168.1.116:9391) Locked, Waiting on external entities 2009-12-02 19:49:56.991392 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2009-12-02 19:49:56.991392 [INFO] mod_dptools.c:2106 Originate Failed. Cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: CALL_REJECTED 2009-12-02 19:49:56.992388 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.116:9391] Other-Leg-Unique-ID: [990cc7dc-e7ea-4b82-850c-9c8c254136f9] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.116:9391] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:57.003383 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [NOTICE] switch_core_session.c:1087 Session 12 (sofia/internal/90964111 at 192.168.1.116:9391) Ended 2009-12-02 19:49:57.004386 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.116:9391 [CS_DESTROY] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9391) State DESTROY 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.116:9391 Standard DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9391) State DESTROY going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_INIT 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.13:5060) State INIT 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA INIT 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.13:5060) State INIT going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.13:5060) State ROUTING 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.13:5060) State ROUTING going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep send 1155 bytes to udp/[192.168.1.13]:5060 at 11:49:57.012792: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522444 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:57.012392 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [calling][0] recv 283 bytes from udp/[192.168.1.13]:5060 at 11:49:57.219574: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 333 bytes from udp/[192.168.1.13]:5060 at 11:49:57.234550: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [proceeding][180] 2009-12-02 19:49:57.233383 [NOTICE] sofia.c:3423 Ring-Ready sofia/internal/90964111 at 192.168.1.13:5060! 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3432 sofia/internal/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 19:49:57.233383 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/internal/PhonerLite at 192.168.1.102! 2009-12-02 19:49:57.233383 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] send 618 bytes to udp/[192.168.1.104]:5060 at 11:49:57.235092: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [early][180] 2009-12-02 19:49:57.233383 [DEBUG] switch_ivr_originate.c:1931 sofia/internal/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 19:49:57.233383 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:57.233383 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/internal/PhonerLite at 192.168.1.102! recv 725 bytes from udp/[192.168.1.13]:5060 at 11:49:59.906802: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 256 v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 376 bytes to udp/[192.168.1.13]:5060 at 11:49:59.907203: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ Max-Forwards: 70 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [ready][200] 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-02 19:49:59.907094 [DEBUG] switch_channel.c:1935 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907094 [NOTICE] sofia.c:3834 Channel [sofia/internal/90964111 at 192.168.1.13:5060] has been answered 2009-12-02 19:49:59.907094 [DEBUG] switch_channel.c:1971 sofia/internal/90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) EXECUTE sofia/internal/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) [1259754599] 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3847 sofia/internal/PhonerLite at 192.168.1.102 receive message [ANSWER] 2009-12-02 19:49:59.907094 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907911 [NOTICE] sofia.c:3847 Channel [sofia/internal/PhonerLite at 192.168.1.102] has been answered 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/internal/90964111 at 192.168.1.13:5060] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_bridge.c:806 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_bridge.c:807 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] send 858 bytes to udp/[192.168.1.104]:5060 at 11:49:59.908359: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ EXECUTE sofia/internal/PhonerLite at 192.168.1.102 get_next_route() 2009-12-02 19:49:59.907911 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [completed][200] EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 19:49:59.907911 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_HIBERNATE] Channel-State-Number: [8] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [answered] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [1259754599907094] Caller-Channel-Progress-Time: [1259754597233383] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] Other-Leg-Unique-ID: [e73fbd5b-a2ca-4685-bc6e-e1f848e8c7e5] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.13:5060] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 ACK Contact: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:59.907911 [DEBUG] mod_dptools.c:752 sofia/internal/PhonerLite at 192.168.1.102 SET [egress_alias]=[9392] 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:355 (sofia/internal/90964111 at 192.168.1.13:5060) State HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] mod_sofia.c:160 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:212 sofia/internal/90964111 at 192.168.1.13:5060 Standard HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:355 (sofia/internal/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep 2009-12-02 19:49:59.907911 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [ready][200] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr.c:1159 sofia/internal/PhonerLite at 192.168.1.102 receive message [MEDIA] 2009-12-02 19:49:59.917033 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.917033 [CRIT] switch_core_io.c:115 sofia/internal/PhonerLite at 192.168.1.102 reading on a session with no media! 2009-12-02 19:49:59.918616 [DEBUG] switch_ivr.c:1174 sofia/internal/90964111 at 192.168.1.13:5060 receive message [MEDIA] 2009-12-02 19:49:59.918616 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] send 955 bytes to udp/[192.168.1.13]:5060 at 11:49:59.922755: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522445 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 33352 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ 2009-12-02 19:49:59.922846 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [calling][0] send 955 bytes to udp/[192.168.1.13]:5060 at 11:50:00.423821: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522445 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 33352 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ recv 337 bytes from udp/[192.168.1.13]:5060 at 11:50:00.444374: ------------------------------------------------------------------------ SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 330 bytes to udp/[192.168.1.13]:5060 at 11:50:00.444561: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:50:00.445918 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [ready][400] 2009-12-02 19:50:00.445918 [NOTICE] sofia.c:3891 Hangup sofia/internal/90964111 at 192.168.1.13:5060 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.445918 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [KILL] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_HANGUP 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.13:5060) State HANGUP 2009-12-02 19:50:00.445918 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.13:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/internal/90964111 at 192.168.1.13:5060 2009-12-02 19:50:00.445918 [NOTICE] switch_ivr_bridge.c:727 Hangup sofia/internal/PhonerLite at 192.168.1.102 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.445918 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/PhonerLite at 192.168.1.102 [KILL] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.13:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.13:5060) State HANGUP going to sleep 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:494 Hangup Command decre_call_stat(203 621 201 256 25 2591585 1): 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.13:5060) State REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.13:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.13:5060) State REPORTING going to sleep 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:1069 Session 13 (sofia/internal/90964111 at 192.168.1.13:5060) Locked, Waiting on external entities 2009-12-02 19:50:00.447416 [ERR] switch_core_io.c:120 sofia/internal/90964111 at 192.168.1.13:5060 has no read codec. 2009-12-02 19:50:00.447416 [DEBUG] switch_ivr_bridge.c:1210 originator uuid 85042f75-de46-4fe5-b8d8-53e4ca0d26e1 is not present 2009-12-02 19:50:00.447416 [NOTICE] switch_core_session.c:1087 Session 13 (sofia/internal/90964111 at 192.168.1.13:5060) Ended 2009-12-02 19:50:00.447416 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.13:5060 [CS_DESTROY] 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.13:5060) State DESTROY 2009-12-02 19:50:00.447416 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.13:5060 Standard DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.13:5060) State DESTROY going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.449755 [INFO] mod_dptools.c:2106 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:1371 Channel is hungup, aborting execution of application: get_next_route 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/PhonerLite at 192.168.1.102) State EXECUTE going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_HANGUP 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/PhonerLite at 192.168.1.102) State HANGUP 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:306 sofia/internal/PhonerLite at 192.168.1.102 Overriding SIP cause 488 with 403 from the other leg 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:338 Channel sofia/internal/PhonerLite at 192.168.1.102 hanging up, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/internal/PhonerLite at 192.168.1.102 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:46 sofia/internal/PhonerLite at 192.168.1.102 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/PhonerLite at 192.168.1.102) State HANGUP going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/PhonerLite at 192.168.1.102) State REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:53 sofia/internal/PhonerLite at 192.168.1.102 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/PhonerLite at 192.168.1.102) State REPORTING going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:1069 Session 10 (sofia/internal/PhonerLite at 192.168.1.102) Locked, Waiting on external entities 2009-12-02 19:50:00.449755 [NOTICE] switch_core_session.c:1087 Session 10 (sofia/internal/PhonerLite at 192.168.1.102) Ended 2009-12-02 19:50:00.449755 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/PhonerLite at 192.168.1.102 [CS_DESTROY] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/PhonerLite at 192.168.1.102) State DESTROY 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:255 sofia/internal/PhonerLite at 192.168.1.102 SOFIA DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:60 sofia/internal/PhonerLite at 192.168.1.102 Standard DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/PhonerLite at 192.168.1.102) State DESTROY going to sleep send 620 bytes to udp/[192.168.1.13]:5060 at 11:50:00.452239: ------------------------------------------------------------------------ BYE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e Max-Forwards: 70 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758964 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ send 641 bytes to udp/[192.168.1.104]:5060 at 11:50:00.452431: ------------------------------------------------------------------------ BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma Max-Forwards: 70 From: ;tag=QQX3yavvBvjrj To: ;tag=2563216860 Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 123758964 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[192.168.1.104]:5060 at 11:50:00.453188: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma From: ;tag=QQX3yavvBvjrj To: ;tag=2563216860 Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 123758964 BYE Contact: User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ recv 337 bytes from udp/[192.168.1.13]:5060 at 11:50:00.530378: ------------------------------------------------------------------------ SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 330 bytes to udp/[192.168.1.13]:5060 at 11:50:00.530547: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 325 bytes from udp/[192.168.1.13]:5060 at 11:50:00.821952: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758964 BYE Contact: Content-Length: 0 ------------------------------------------------------------------------ From mike at jerris.com Tue Dec 1 20:26:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 23:26:22 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> Message-ID: <944399F2-5F9F-437D-B57B-5DA4FC472B76@jerris.com> That is not a freeswitch log and the call path you describe is not what was in the previous freeswitch log you posted. Please post the complete freeswitch debug log with siptrace enabled for that callflow you described. On Dec 1, 2009, at 11:07 PM, Juan Backson wrote: > Hi Mike, > > > Here is a very strange SIP outgoing INVITES from freeswitch: > > The call path is this: > > 192.168.1.104 (phonerlite) -> 192.168.1.102 ( freeswitch ) > 192.168.1.102 -> 192.168.1.116 (sipp gives back 403) > 192.168.1.102 -> 192.168.1.13 ( phone ) > > The first INVITE to 192.168.1.13 has the right c= and o= ( both is > pointing to 192.168.1.104). But the for some unknown reason, > Freeswitch sends INVITES again. But in the INVITE resend, the o = > becomes fs's ip. > > I have no idea. This is only bypass_meida. > > Attached is the fs log. > > > ngrep -q -p -W byline port 5060 > interface: eth0 (192.168.1.0/255.255.255.0) > filter: (ip) and ( port 5060 ) > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > INVITE sip:90964111 at 192.168.1.102 SIP/2.0. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport. > From: ;tag=2563216860. > To: . > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > Content-Type: application/sdp. > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE. > Max-Forwards: 70. > Supported: 100rel, replaces. > User-Agent: SIPPER for PhonerLite. > Content-Length: 396. > . > v=0. > o=- 478760567 0 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=sendrecv. > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: . > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9390 > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > Max-Forwards: 69. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: . > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 401. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 523740055483911509 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.116:9390 -> 192.168.1.102:5060 > SIP/2.0 403 Because. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: ;tag=6. > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9390 > ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > Max-Forwards: 69. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: ;tag=6. > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9391 > INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > Max-Forwards: 69. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: . > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 402. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 5392773558290384508 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.116:9391 -> 192.168.1.102:5060 > SIP/2.0 403 Because. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: ;tag=6. > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9391 > ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > Max-Forwards: 69. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: ;tag=6. > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: . > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 402. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522444 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: . > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Supported: replaces. > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, > PRACK, UPDATE, MESSAGE. > Content-Type: application/sdp. > Content-Length: 256. > . > v=0. > o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. > s=A conversation. > c=IN IP4 192.168.1.13. > t=0 0. > m=audio 10098 RTP/AVP 8 0 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:9 G722/16000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > ACK sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ. > Max-Forwards: 70. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 207. > . > v=0. > o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. > s=A conversation. > c=IN IP4 192.168.1.13. > t=0 0. > m=audio 10098 RTP/AVP 8 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK00b67e2664ddde1180d3001a805656a5;rport. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 ACK. > Contact: . > Max-Forwards: 70. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 223. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.102. > t=0 0. > m=audio 33352 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > > > U 192.168.1.104:43488 -> 192.168.1.102:5060 > . > . > .............. > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 223. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.102. > t=0 0. > m=audio 33352 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 400 Bad Request. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > ACK sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > BYE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e. > Max-Forwards: 70. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758964 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma. > Max-Forwards: 70. > From: ;tag=QQX3yavvBvjrj. > To: ;tag=2563216860. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 123758964 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". > Content-Length: 0. > . > > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma. > From: ;tag=QQX3yavvBvjrj. > To: ;tag=2563216860. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 123758964 BYE. > Contact: . > User-Agent: SIPPER for PhonerLite. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 400 Bad Request. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > Content-Length: 0. > . > > > > On Wed, Dec 2, 2009 at 11:26 AM, Michael Jerris > wrote: > how are you sending both invites here? can you explain the full > call path and how you are originating these calls? > > > On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > >> Hi, >> >> I also did try to set?only bypass_media, but it still does not wor >> k??freeswitch still modifies the c= line, causing the call to fa >> il. >> >> Could someone please help? >> >> >> send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: >> >> --- >> --------------------------------------------------------------------- >> INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> Max-Forwards: 69 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 402 >> Remote-Party-ID: "PhonerLite" >> ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change CS_INIT -> >> CS_ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to >> sleep >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/ >> external_1/90964111 at 192.168.1.13:5060 SOFIA ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going >> to sleep >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA >> going to sleep >> recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [proceeding] >> [180] >> 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/ >> external_1/90964111 at 192.168.1.13:5060! >> 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ >> PhonerLite at 192.168.1.102 receive message [RINGING] >> 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready >> sofia/external_1/PhonerLite at 192.168.1.102! >> 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 >> sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] >> 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [early][180] >> 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 >> Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! >> recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> Supported: replaces >> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, >> PRACK, UPDATE, MESSAGE >> Content-Type: application/sdp >> Content-Length: 256 >> >> v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> --- >> --------------------------------------------------------------------- >> send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: >> >> --- >> --------------------------------------------------------------------- >> ACK sip:192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp >> Max-Forwards: 70 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 ACK >> Contact: >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [ready][200] >> 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/ >> external_1/90964111 at 192.168.1.13:5060] has been answered >> 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/ >> external_1/90964111 at 192.168.1.13:5060 execute on answer: >> incre_call_stat(203 621 201 256 25 2591585 1) >> EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1 >> ) >> >> 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 >> Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered >> 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 >> Originate Resulted in Success: [sofia/ >> external_1/90964111 at 192.168.1.13:5060] >> send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N2009-12-02 >> 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ >> PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE >> >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media= >> [true] >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 >> originate_disposition=[SUCCESS] >> Content-Type: application/sdp >> Content-Disposition: session >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> Content-Length: 207 >> >> v=0 >> 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change >> CS_CONSUME_MEDIA -> CS_HIBERNATE >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> m=audio 10096 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/ >> external_1/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 >> sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going >> to sleep >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() >> 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to >> get next route... >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() >> 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_HIBERNATE] >> Channel-State-Number: [8] >> Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] >> Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [answered] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259751417450336] >> Caller-Channel-Created-Time: [1259751417450336] >> Caller-Channel-Answered-Time: [1259751419381828] >> Caller-Channel-Progress-Time: [1259751417736234] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> Other-Leg-Username: [PhonerLite] >> Other-Leg-Caller-ID-Name: [PhonerLite] >> Other-Leg-Caller-ID-Number: [PhonerLite] >> Other-Leg-Network-Addr: [192.168.1.104] >> Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] >> Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] >> Other-Leg-Source: [mod_sofia] >> Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] >> Other-Leg-Screen-Bit: [true] >> Other-Leg-Privacy-Hide-Name: [false] >> Other-Leg-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [2454193703] >> variable_sofia_profile_name: [external_1] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [000BE0BD-5CDD- >> DE11-80D1-001A805656A5 at 192.168.1.104] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 794697697 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> variable_proto_specific_hangup_cause: [sip:403] >> variable_sip_hangup_phrase: [Because] >> variable_bypass_media: [true] >> variable_success_bridge: [true] >> variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] >> variable_switch_m_sdp: [v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> ] >> variable_endpoint_disposition: [ANSWER] >> variable_originate_disposition: [SUCCESS] >> variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] >> variable_current_application: [info] >> >> recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: >> >> --- >> --------------------------------------------------------------------- >> ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 >> sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]= >> [90964111] >> >> Contact: >> Max-Forwards: 70 >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/ >> external_1/90964111 at 192.168.1.116:9392) >> 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/ >> external_1/PhonerLite at 192.168.1.102 receive message [MEDIA] >> 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/ >> external_1/PhonerLite at 192.168.1.102 reading on a session with no >> media! >> 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [completed][200] >> 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [ready][200] >> 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/ >> external_1/90964111 at 192.168.1.13:5060 receive message [MEDIA] >> 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: >> >> --- >> --------------------------------------------------------------------- >> INVITE sip:192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj >> Max-Forwards: 69 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757373 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 223 >> Remote-Party-ID: "PhonerLite" >> ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 30632 RTP/AVP 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] >> send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: >> >> --- >> --------------------------------------------------------------------- >> >> On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale > > wrote: >> yes he did you can see it in his trace. >> you can not use both of them together...... >> >> >> >> On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris >> wrote: >> The only way this would happen would be if this is set to proxy >> media not bypass. Are you setting both? >> >> Mike >> >> On Dec 1, 2009, at 10:08 AM, Juan Backson >> wrote: >> >>> In the following trace, 102 is FS IP, 104 is calling party and >>> 13 is called party. >>> >>> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >>> >>> Any idea why? >>> >>> >>> freeswitch at localhost.localdomain> recv 951 bytes from udp/ >>> [192.168.1.104]:5060 at 22:56:33.782715: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >>> Via: SIP/2.0/UDP >>> 192.168.1.104: >>> 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >>> From: ;tag=786224322 >>> To: >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> Contact: >>> Content-Type: application/sdp >>> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, >>> MESSAGE, UPDATE >>> Max-Forwards: 70 >>> Supported: 100rel, replaces >>> User-Agent: SIPPER for PhonerLite >>> Content-Length: 397 >>> >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=sendrecv >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP >>> 192.168.1.104: >>> 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >>> From: ;tag=786224322 >>> To: >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Content-Length: 0 >>> >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New >>> Channel sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a- >>> ee3b-40d4-910d-3b1579f9a273] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ >>> internal/PhonerLite at 192.168.1.102 entering state [received][100] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> >>> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >>> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >>> Channel-State: [CS_EXECUTE] >>> Channel-State-Number: [4] >>> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Call-Direction: [inbound] >>> Presence-Call-Direction: [inbound] >>> Answer-State: [ringing] >>> Caller-Username: [PhonerLite] >>> Caller-Dialplan: [class4] >>> Caller-Caller-ID-Name: [PhonerLite] >>> Caller-Caller-ID-Number: [PhonerLite] >>> Caller-Network-Addr: [192.168.1.104] >>> Caller-Destination-Number: [90964111] >>> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Caller-Source: [mod_sofia] >>> Caller-Context: [default] >>> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Caller-Profile-Index: [1] >>> Caller-Profile-Created-Time: [1259708193783162] >>> Caller-Channel-Created-Time: [1259708193783162] >>> Caller-Channel-Answered-Time: [0] >>> Caller-Channel-Progress-Time: [0] >>> Caller-Channel-Progress-Media-Time: [0] >>> Caller-Channel-Hangup-Time: [0] >>> Caller-Channel-Transfer-Time: [0] >>> Caller-Screen-Bit: [true] >>> Caller-Privacy-Hide-Name: [false] >>> Caller-Privacy-Hide-Number: [false] >>> variable_sip_received_ip: [192.168.1.104] >>> variable_sip_received_port: [5060] >>> variable_sip_via_protocol: [udp] >>> variable_sip_from_user: [PhonerLite] >>> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >>> variable_sip_from_host: [192.168.1.102] >>> variable_sip_from_user_stripped: [PhonerLite] >>> variable_sip_from_tag: [786224322] >>> variable_sofia_profile_name: [internal] >>> variable_sip_req_user: [90964111] >>> variable_sip_req_uri: [90964111 at 192.168.1.102] >>> variable_sip_req_host: [192.168.1.102] >>> variable_sip_to_user: [90964111] >>> variable_sip_to_uri: [90964111 at 192.168.1.102] >>> variable_sip_to_host: [192.168.1.102] >>> variable_sip_contact_port: [5060] >>> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >>> variable_sip_contact_host: [192.168.1.104] >>> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >>> variable_sip_call_id: [003C8E1B-F8DC-DE11- >>> A853-001A805656A5 at 192.168.1.104] >>> variable_sip_user_agent: [SIPPER for PhonerLite] >>> variable_sip_via_host: [192.168.1.104] >>> variable_sip_via_port: [5060] >>> variable_bypass_media: [true] >>> variable_proxy_media: [true] >>> variable_sip_via_rport: [5060] >>> variable_max_forwards: [70] >>> variable_switch_r_sdp: [v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> ] >>> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >>> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >>> variable_effective_caller_id_number: [PhonerLite] >>> variable_effective_caller_id_name: [PhonerLite] >>> variable_>> variable_routing_digit: [90964111] >>> variable_continue_on_fail: [true] >>> variable_hangup_after_bridge: [true] >>> variable_sip_contact_user: [PhonerLite] >>> >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/ >>> internal/90964111 at 192.168.1.116:9390 Patched SDP >>> --- >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> +++ >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >>> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to >>> sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State Change >>> CS_ROUTING -> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going >>> to sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> going to sleep >>> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >>> Max-Forwards: 69 >>> From: "PhonerLite" >>> ;tag=8tH6Xjt2XaU9F >>> To: >>> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >>> CSeq: 123735760 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 404 >>> Remote-Party-ID: "PhonerLite" >>> ;party=calling;screen=yes;privacy=off >>> >>> v=0 >>> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/2a6656ad/attachment-0001.html From max.clark at gmail.com Tue Dec 1 20:03:39 2009 From: max.clark at gmail.com (Max Clark) Date: Tue, 01 Dec 2009 20:03:39 -0800 Subject: [Freeswitch-users] OSP Interop w/ Trans Nexus In-Reply-To: <191c3a030810310758l4cf91940v39f30e611fd05e2c@mail.gmail.com> References: <200810311312.m9VDCoQM031972@omr12.networksolutionsemail.com> <191c3a030810310758l4cf91940v39f30e611fd05e2c@mail.gmail.com> Message-ID: Hi all, Did anything ever progress with this? Is there an option for OSP in FreeSWITCH? Thanks, Max On 10/31/08 7:58 AM, Anthony Minessale wrote: > We're here all the time if you want to collaborate on it. > We have 100+ users and developers in our irc channel and on this list so > it should not be an issue. > I'm sure we can find a few volunteers for testing. > > > On Fri, Oct 31, 2008 at 8:14 AM, Jim Dalton > > wrote: > > TransNexus would be glad to contribute the effort to add support for > the ETSI OSP protocol to Freeswitch if there is interest from the > community. Since we are not familiar with FreeSwitch, we will need > to collaborate with a FreeSwitch developer to understand how the OSP > client library (http://sourceforge.net/projects/osp-toolkit/) should > be integrated with Freeswitch. We will also need a user who can > perform beta testing on the Freeswitch OSP implementation. > Jim Dalton > VoIP Routing, Accounting, Security > 1.404.526.6053 > www.TransNexus.com From devel at thom.fr.eu.org Wed Dec 2 00:12:09 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 09:12:09 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> Message-ID: <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [2] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [4] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [6] http://www.freeswitch.org [7] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [8] ClueCon http://www.cluecon.com/ [9] Twitter: http://twitter.com/FreeSWITCH_wire [10] AIM: anthm MSN:anthony_minessale at hotmail.com [11] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [12] IRC: irc.freenode.net [13] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [14] iax:guest at conference.freeswitch.org/888 [15] googletalk:conf+888 at conference.freeswitch.org [16] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [17] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [19] http://www.freeswitch.org [20] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [21] ClueCon http://www.cluecon.com/ [22] Twitter: http://twitter.com/FreeSWITCH_wire [23] AIM: anthm MSN:anthony_minessale at hotmail.com [24] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [25] IRC: irc.freenode.net [26] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [27] iax:guest at conference.freeswitch.org/888 [28] googletalk:conf+888 at conference.freeswitch.org [29] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] http://files.freeswitch.org [3] mailto:devel at thom.fr.eu.org [4] mailto:FreeSWITCH-users at lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6] http://lists.freeswitch.org/mailman/options/freeswitch-users [7] http://www.freeswitch.org [8] http://www.freeswitch.org/ [9] http://www.cluecon.com/ [10] http://twitter.com/FreeSWITCH_wire [11] mailto:MSN%3Aanthony_minessale at hotmail.com [12] mailto:PAYPAL%3Aanthony.minessale at gmail.com [13] http://irc.freenode.net [14] mailto:sip%3A888 at conference.freeswitch.org [15] http://iax:guest at conference.freeswitch.org/888 [16] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [17] mailto:FreeSWITCH-users at lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] http://www.freeswitch.org/ [22] http://www.cluecon.com/ [23] http://twitter.com/FreeSWITCH_wire [24] mailto:MSN%3Aanthony_minessale at hotmail.com [25] mailto:PAYPAL%3Aanthony.minessale at gmail.com [26] http://irc.freenode.net [27] mailto:sip%3A888 at conference.freeswitch.org [28] http://iax:guest at conference.freeswitch.org/888 [29] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/77d63b4b/attachment.html From abeka at greatiam.com Wed Dec 2 00:16:49 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 08:16:49 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers Message-ID: <4B162271.1010306@greatiam.com> Hello I am experimenting with FS and would like to know how to connect two independent servers with user on one beinng able to call users on the other. Do I set each server to be the gateway of the corresponding one ? Pardon me if this has already benn dealt with. My search has drawn a blank Thanks. From jbarou at sqli.com Wed Dec 2 01:54:05 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Wed, 2 Dec 2009 10:54:05 +0100 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> Message-ID: <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> Any ideas ? Thanks 2009/11/30 Jonathan Barou > My version is FreeSWITCH Version 1.0.trunk (15691M) > > http://jira.freeswitch.org/browse/FSBUILD-213 > > Thanks you. > > 2009/11/27 Anthony Minessale > > by latest do you mean SVN trunk? >> >> Can you issue the command "sofia profile internal siptrace on" before >> capturing your trace and post the results >> to http://pastebin.freeswitch.org or open a jira >> http://jira.freeswitch.org on the issue and attach the log after you >> create the issue ticket, don't include it in the mailing list. >> >> >> On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: >> >>> Hi everybody, >>> >>> I'm actually using the lastest version of Freeswitch, I have a problem. I >>> have a trunk SIP with my PABX. >>> >>> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >>> 2. one Alcatel IpTouch 4028 with number 987 >>> (on PABX) >>> 3. one Siemens Gigaset A580 IP with number >>> 8401 (on Freeswitch) >>> >>> >>> *The first test* is to say to the phone 2 to transfer all the call to >>> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >>> phone 3 is ringing and it's work. I have that in the log : >>> >>> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >>> early media >>> >>> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> >>> 10.33.69.246 port 32000 codec: 8 ms: 90 >>> >>> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer >>> [soft] 720 bytes per 90ms >>> >>> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >>> >>> v=0 >>> >>> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >>> >>> s=FreeSWITCH >>> >>> c=IN IP4 10.33.169.92 >>> >>> t=0 0 >>> >>> m=audio 23054 RTP/AVP 8 106 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:106 telephone-event/8000 >>> >>> a=fmtp:106 0-16 >>> >>> a=silenceSupp:off - - - - >>> >>> a=ptime:90 >>> >>> a=sendrecv >>> >>> >>> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >>> sofia/internal/368 at 10.33.69.246! >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>> 368 at 10.33.69.246 skipping state [early][183] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >>> sofia/internal/368 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>> Activation Success L16 at 8000hz 1 channel 90ms >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >>> Ringback Tone [%(2000,4000,440.0,480.0)] >>> >>> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >>> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>> >>> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >>> confirmed. >>> >>> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >>> Buffer at 1440 bytes vs 81 >>> >>> >>> >>> *The Second Tes*t is to say to the phone 1 to transfer all the call to >>> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >>> one time and after it hangup. I have that in the log : >>> >>> >>> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >>> early media >>> >>> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> >>> 10.33.69.144 port 32000 codec: 8 ms: 90 >>> >>> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer >>> [soft] 720 bytes per 90ms >>> >>> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >>> >>> v=0 >>> >>> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >>> >>> s=FreeSWITCH >>> >>> c=IN IP4 10.33.169.92 >>> >>> t=0 0 >>> >>> m=audio 27732 RTP/AVP 8 106 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:106 telephone-event/8000 >>> >>> a=fmtp:106 0-16 >>> >>> a=silenceSupp:off - - - - >>> >>> a=ptime:90 >>> >>> a=sendrecv >>> >>> >>> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >>> sofia/internal/987 at 10.33.69.246! >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>> 987 at 10.33.69.246 skipping state [early][183] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>> Activation Success L16 at 8000hz 1 channel 90ms >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >>> Ringback Tone [%(2000,4000,440.0,480.0)] >>> >>> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >>> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >>> 987 at 10.33.69.246 entering state [terminated][487] >>> >>> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >>> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >>> sofia/internal/987 at 10.33.69.246 [KILL] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread >>> mismatch skipping state handler. >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >>> original codec. >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >>> [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >>> other leg >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>> sofia/internal/sip:8401 at 10.33.170.231:5060 hanging up, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >>> sofia/internal/sip:8401 at 10.33.170.231:5060 >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >>> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >>> sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING >>> -> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external >>> entities >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread >>> mismatch skipping state handler. >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to >>> sleep >>> >>> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot >>> create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >>> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >>> Cause: ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/987 at 10.33.69.246) State HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>> sofia/internal/987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/987 at 10.33.69.246) State REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> >>> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >>> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 >>> (sofia/internal/987 at 10.33.69.246) Ended >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/987 at 10.33.69.246) State DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>> 987 at 10.33.69.246 SOFIA DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/987 at 10.33.69.246 Standard DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >>> >>> Finally when I tried to call the phone 3 with the phone 1 it's working, >>> and not when I want to call the phone 3 with the phone 2, like just before, >>> it's ringing just one time and hangup. >>> >>> >>> Thanks you. >>> >>> >>> Best Regards >>> >>> -- >>> John >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jonathan BAROU > Groupe SQLI - CRCI > > 0472405368 > jbarou at sqli.com > > 1, place Verrazzano > 69258 LYON CEDEX 09 > > -- Jonathan BAROU Groupe SQLI - CRCI 0472405368 jbarou at sqli.com 1, place Verrazzano 69258 LYON CEDEX 09 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/2109ac3f/attachment-0001.html From sharad at coraltele.com Wed Dec 2 02:02:43 2009 From: sharad at coraltele.com (sharad) Date: Wed, 2 Dec 2009 15:32:43 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 42, Issue 12 References: Message-ID: <000f01ca7336$97bb1950$0c04a8c0@compaq77db609e> Hello We also faced the similar issue. Actually it is caused bacause hold on music files are missing. either you save all the music files or configure your dialplan accordingly. Sharad , Coral Telecom, India ----- Original Message ----- From: To: Sent: Wednesday, December 02, 2009 3:24 PM Subject: FreeSWITCH-users Digest, Vol 42, Issue 12 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > > 1. Re: OSP Interop w/ Trans Nexus (Max Clark) > 2. Re: CLIP on FXS channels with mod_openzap (Fran?ois Legal) > 3. Bridging/Connecting Freeswitch servers (Otis) > 4. Re: Transfer Problem (Jonathan Barou) > -------------------------------------------------------------------------------- > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From devel at thom.fr.eu.org Wed Dec 2 02:57:12 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 11:57:12 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0229b88f/attachment.html From kond at nstel.ru Wed Dec 2 00:34:42 2009 From: kond at nstel.ru (Nikolay Kondratyev) Date: Wed, 2 Dec 2009 11:34:42 +0300 Subject: [Freeswitch-users] call barge in Message-ID: <20091202083449.271C2116CD@mail.nstel.ru> Hi all, I'm evaluating FS for our organization. I must fulfill the following requirements: 1. Call recording: All (or selected) calls to the secretary must be recorded. 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes "emergency" (for example, an extension with emergency prefix) call to one of these subscribers -> Secretary barges in the established call (conference). 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. Can anybody please advise if this is possible with FS? If yes, is it just a configuration task, or additional programming will be needed? Thanks in advance, Nikolay. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/125993b7/attachment.html From frank at carmickle.com Wed Dec 2 05:45:27 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 08:45:27 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Message-ID: <20091202134526.GR31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > > > Hello, > > I created an extension in my dialplan so that when an incoming > call arrives, it rings a group of lines and then fallback to the voicemail > if no line is answered. > > I wanted then that when voicemail starts, the > calling party could dial some numbers to fetch the voicemail. I used > bind_meta_app for this. My problem is, when using bind_meta_app, the > voicemail continues, and I sometimes experience freeswitch hanging after > the call is over, depending on when the bind_meta_app is activated. The action your looking for is what is bound to "*" in the default voicemail config. Look at autoload_configs/voicemail.conf.xml HTH --FC From frank at carmickle.com Wed Dec 2 06:01:55 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 09:01:55 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B162271.1010306@greatiam.com> References: <4B162271.1010306@greatiam.com> Message-ID: <20091202140155.GS31924@base.carmickle.com> On Wed, Dec 02, Otis wrote: > Hello > > I am experimenting with FS and would like to know how to connect two > independent servers with user on one beinng able to call users on the > other. Do I set each server to be the gateway of the corresponding one ? You can if you need them to authenticate to eachother. You have to decide on what you need. Do you not want extensions reachable from the public context? If not then you can do what I do. You can certainly put an ipv4 address in instead of the mangled ipv6 that's in this example. Then create an extension that matches on the extensions on the other machine and bridge them to the correct hostname and port. If you just want all the extensions reachable from the public context then do something like this in your dialplan/public.xml There are yet other ways to get this done. HTH --FC From erandr-junk at usa.net Wed Dec 2 06:47:47 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 06:47:47 -0800 (PST) Subject: [Freeswitch-users] Best way to run originate calls through dial plan Message-ID: <26610094.post@talk.nabble.com> What would be the best way of making originate() run call through a dial plan (compared to directly going to a specified VOIP gateway). Would it be loopbacks, i.e. smth like this? /opt/freeswitch/bin/fs_cli -x "originate {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js zzzzzzzzzz)'" The idea of this is that originate() sets up the first call, then webcall.js plays back a WAV, and bridges the first call with the second one (also set up via loopback). Thanks! -- View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26610094.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Wed Dec 2 07:23:53 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 2 Dec 2009 17:23:53 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name Message-ID: Hello, We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI there is a Nortel with Q.Sig. After a lot of configuration trials I've managed to set it to send back the connected name over the SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the connected name and then the Cisco adds it as a Remote-Party-ID). However, I did not save it and a power outage cleared this config. In my age I don't remember what I've done... Anyone knows the correct config? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/bb473cd7/attachment.html From devel at thom.fr.eu.org Wed Dec 2 07:28:29 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 16:28:29 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202134526.GR31924@base.carmickle.com> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> Message-ID: <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> On Wed, 2 Dec 2009 08:45:27 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> >> >> Hello, >> >> I created an extension in my dialplan so that when an incoming >> call arrives, it rings a group of lines and then fallback to the >> voicemail >> if no line is answered. >> >> I wanted then that when voicemail starts, the >> calling party could dial some numbers to fetch the voicemail. I used >> bind_meta_app for this. My problem is, when using bind_meta_app, the >> voicemail continues, and I sometimes experience freeswitch hanging after >> the call is over, depending on when the bind_meta_app is activated. > > The action your looking for is what is bound to "*" in the default > voicemail config. Look at autoload_configs/voicemail.conf.xml > > > > HTH > --FC > I tried to remove the bind_meta_app from the dialplan, call the extension then press * when the greeting message starts, but it did not bring the voicemail prompt for my id and password. Fran?ois > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at freeswitch.org Wed Dec 2 07:46:04 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 2 Dec 2009 13:46:04 -0200 Subject: [Freeswitch-users] CDR records In-Reply-To: <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> Message-ID: What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions. JM On Tue, Dec 1, 2009 at 3:31 PM, Michael Collins wrote: > > > On Sun, Nov 29, 2009 at 10:06 AM, Pusk?s Zsolt wrote: > >> Hi Guys! >> >> I'm using the latest svn (15711) with the default xml config. Only >> modified >> cdr_csv.conf.xml the line to > name="legs" >> value="ab"/> >> >> Here is what i do: >> >> 1. 1000 calls 1001 (1001 answers the call) >> 2. 1001 do blind transfer to 1002 (using *1) >> 3. 1001 hangs up >> 4. 1002 answers the call >> 5. 1002 and 1000 hangs up >> >> 3 cdr records are generated (simplified): >> >> from,to,start,duration >> "1000" "1001" "2009-11-29 15:21:53" "53" "50" >> "1000" "1002" "2009-11-29 15:21:53" "79" "76" >> "1000" "1002" "2009-11-29 15:22:46" "26" "23" >> >> As you can see the second cdr is incorrect because 1000 doesn't speak with >> 1002 for 76 second. >> >> Is this a normal ? Is it possible to make only 2 record ? >> >> You may want to turn on mod_xml_curl and look at XML CDRs, comparing them > to the corresponding CSV files. That should help you figure out why the > values in the CSV are what they are. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/fbee6871/attachment-0001.html From frank at carmickle.com Wed Dec 2 08:00:11 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 11:00:11 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> Message-ID: <20091202160011.GU31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: Snip... > > voicemail config. Look at autoload_configs/voicemail.conf.xml > > > > > > > > HTH > > --FC > > > > I tried to remove the bind_meta_app from the dialplan, call the extension > then press * when the greeting message starts, but it did not bring the > voicemail prompt for my id and password. Did you check your voicemail config as I pointed out? autoload_configs/voicemail.conf.xml should have And what exactly do you mean by "Remote fetching of voicemail?" --FC From kristian.kielhofner at gmail.com Wed Dec 2 08:35:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 2 Dec 2009 11:35:40 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> Message-ID: <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> As always, you are correct. The scenario now is: - If the caller places the callee on hold, the callee will get hold music - If the callee places the caller on hold, the caller will not get hold music I've uploaded a fresh pastebin here: http://pastebin.freeswitch.org/11356 On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale wrote: > results cant possibly be the same > there is not even any broadcast involved in uuid_transfer ? > > you need to attach a console trace with debug log up > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From john_platts at hotmail.com Wed Dec 2 08:39:24 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 2 Dec 2009 10:39:24 -0600 Subject: [Freeswitch-users] Update to MODENDP-272 Message-ID: I have uploaded the dialplan and JavaScript files used to process calls to MODENDP-272. I have even done a make current to revision 15755, and the blind transfer is still failing. _________________________________________________________________ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 From devel at thom.fr.eu.org Wed Dec 2 08:49:46 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 17:49:46 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202160011.GU31924@base.carmickle.com> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> <20091202160011.GU31924@base.carmickle.com> Message-ID: <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> I did check (and modify as my voicemail extension name is not vmain) the voicemail.conf.xml, and vmain-key is *. What I mean by remote fetching of voicemail, is that someone may dial in, either from inside (via FXS or even SIP) or outside (via FXO), then when reaching the voice mail to leave a message, he could dial some specific digit (or digits) to reach the voicemail login and fetch his voice mails. I can do this using bind_meta_app (it is already working), but then I need to terminate the extension when invoking the meta_app, otherwise freeswitch may sometimes hang if the meta app is called after the "leave a message" voicemail tone. Fran?ois On Wed, 2 Dec 2009 11:00:11 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: > Snip... >> > voicemail config. Look at autoload_configs/voicemail.conf.xml >> > >> > >> > >> > HTH >> > --FC >> > >> >> I tried to remove the bind_meta_app from the dialplan, call the extension >> then press * when the greeting message starts, but it did not bring the >> voicemail prompt for my id and password. > > Did you check your voicemail config as I pointed out? > autoload_configs/voicemail.conf.xml should have > > > > And what exactly do you mean by "Remote fetching of voicemail?" > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From frank at carmickle.com Wed Dec 2 09:16:14 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 12:16:14 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> <20091202160011.GU31924@base.carmickle.com> <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> Message-ID: <20091202171613.GV31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > I did check (and modify as my voicemail extension name is not vmain) the > voicemail.conf.xml, and vmain-key is *. > > What I mean by remote fetching of voicemail, is that someone may dial in, > either from inside (via FXS or even SIP) or outside (via FXO), then when > reaching the voice mail to leave a message, he could dial some specific > digit (or digits) to reach the voicemail login and fetch his voice mails. > > I can do this using bind_meta_app (it is already working), but then I need > to terminate the extension when invoking the meta_app, otherwise freeswitch > may sometimes hang if the meta app is called after the "leave a message" > voicemail tone. Alright. I missed what vmain actually does in the voicemail config. It actually calls the extension named vmain in the dialplan. So if you don't have this then you will need to have one. Thanks for asking this question because my voicemail auth was broken and I didn't even know it! I fixed it and a working extension for vmain can look like this. HTH --FC From shiyanov at gmail.com Wed Dec 2 09:21:53 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 2 Dec 2009 20:21:53 +0300 Subject: [Freeswitch-users] call barge in In-Reply-To: <20091202083449.271C2116CD@mail.nstel.ru> References: <20091202083449.271C2116CD@mail.nstel.ru> Message-ID: 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Here is a bunch of code, search there ''barge" Artem On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote: > Hi all, > > > > I?m evaluating FS for our organization. > > I must fulfill the following requirements: > > 1. Call recording: All (or selected) calls to the secretary must be > recorded. > > 2. Call barge in: Assume that two subscribers are talking to each other. > Secretary makes ?emergency? (for example, an extension with emergency > prefix) call to one of these subscribers -> Secretary barges in the > established call (conference). > > 3. Call drop when emergency call arrives: the same as above, but > established call is dropped end emergency call is established. > > > > Can anybody please advise if this is possible with FS? > > If yes, is it just a configuration task, or additional programming will be > needed? > > > > Thanks in advance, > > Nikolay. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/c5807777/attachment.html From abeka at greatiam.com Wed Dec 2 09:29:26 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 17:29:26 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202140155.GS31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> Message-ID: <4B16A3F6.6000702@greatiam.com> Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > >> Hello >> >> I am experimenting with FS and would like to know how to connect two >> independent servers with user on one beinng able to call users on the >> other. Do I set each server to be the gateway of the corresponding one ? >> > > You can if you need them to authenticate to eachother. You have to decide on what you need. Do you not want extensions reachable from the public context? If not then you can do what I do. > > > > > > > > > You can certainly put an ipv4 address in instead of the mangled ipv6 that's in this example. > > Then create an extension that matches on the extensions on the other machine and bridge them to the correct hostname and port. > > If you just want all the extensions reachable from the public context then do something like this in your dialplan/public.xml > > > > > > > > There are yet other ways to get this done. > > HTH > --FC > > > Thanks. I would like all extensions on say server A to be contactable by those on server B and vice versa. From msc at freeswitch.org Wed Dec 2 09:35:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:35:06 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> Message-ID: <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> 2009/12/2 Jo?o Mesquita > What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions. > > JM > > Thanks for catching my typo! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/327f34ad/attachment.html From devel at thom.fr.eu.org Wed Dec 2 09:43:32 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 18:43:32 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: No, my voicemail extension (I have 2 actually) is called vmain_unregistered_user, so in voicemail.conf.xml I have : But still (and I don't even know if I'm using it the right way), I would expect that when the voicemail greeting starts, I could press * on the phone to call the vmain_unregistered_user extension in my dialplan, but that never happens. What I want, is when someone dials in (this is usefull only from outside so via FXO line) his extension (or any other, it's not important) that when he reach the voicemail to leave a message, override the "leave a message" voicemail and enter the "check my messages" voicemail (with an authentication step). I'm not sure this is clear. Fran?ois On Wed, 2 Dec 2009 12:16:14 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> I did check (and modify as my voicemail extension name is not vmain) the >> voicemail.conf.xml, and vmain-key is *. >> >> What I mean by remote fetching of voicemail, is that someone may dial in, >> either from inside (via FXS or even SIP) or outside (via FXO), then when >> reaching the voice mail to leave a message, he could dial some specific >> digit (or digits) to reach the voicemail login and fetch his voice mails. >> >> I can do this using bind_meta_app (it is already working), but then I >> need >> to terminate the extension when invoking the meta_app, otherwise >> freeswitch >> may sometimes hang if the meta app is called after the "leave a message" >> voicemail tone. > > Alright. I missed what vmain actually does in the voicemail config. It > actually calls the extension named vmain in the dialplan. So if you don't > have this then you will need to have one. Thanks for asking this question > because my voicemail auth was broken and I didn't even know it! I fixed it > and a working extension for vmain can look like this. > > > expression="^vmain$|^voicemail$|^\*98$|^\*86$"> > > > > > > > > HTH > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 2 09:47:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:47:49 -0800 Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <26610094.post@talk.nabble.com> References: <26610094.post@talk.nabble.com> Message-ID: <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: > > What would be the best way of making originate() run call through a dial > plan > (compared to directly going to a specified VOIP gateway). Would it be > loopbacks, i.e. smth like this? > > /opt/freeswitch/bin/fs_cli -x "originate > > {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML > '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js zzzzzzzzzz)'" > > The idea of this is that originate() sets up the first call, then > webcall.js > plays back a WAV, and bridges the first call with the second one (also set > up via loopback). > > Could you describe the problem that you're trying to solve? That would make it easier to know if what you've come up with is the best solution. How many calls per second were you wanting to generate with this setup? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/5b63db3d/attachment.html From msc at freeswitch.org Wed Dec 2 09:50:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:50:03 -0800 Subject: [Freeswitch-users] Update to MODENDP-272 In-Reply-To: References: Message-ID: <87f2f3b90912020950g597b4217r6b198175044539f0@mail.gmail.com> On Wed, Dec 2, 2009 at 8:39 AM, John Platts wrote: > > I have uploaded the dialplan and JavaScript files used to process calls to > MODENDP-272. I have even done a make current to revision 15755, and the > blind transfer is still failing. > > John, Thanks for keeping the guys in the loop. Just a quick note: when you make updates to JIRA cases that you've opened the devs will see it so there's no need to send an email to the users list. :) Hold tight and the devs will check it out as soon as they can. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/4c89b033/attachment.html From dlaperle at rsslex.com Wed Dec 2 07:00:25 2009 From: dlaperle at rsslex.com (David Laperle) Date: Wed, 02 Dec 2009 10:00:25 -0500 Subject: [Freeswitch-users] Dictation System Message-ID: <1259766025.1980.11.camel@dlaplap> Hi Freeswitch users, i'm new into the PBX world. I just installed FreeSwitch and made work great, but one of my goal with the PBX system is to use it as a dictation system. We were using Callweaver, and there's a Dictation module for CW and one for Asterisk, but i can't find one for FreeSwitch so far. Is there anything in the trunk that i could find or any work in progress? I'm willing to develop a bit to help the work in progress, i have the programming knowledge but not the VOIP/PBX knowledge, so a work in progress could be enough for me to start with and complete the work! If any of you have an idea or a hint for me i would be very grateful! Thanks a lot, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. Bien que ce courriel ait ?t? trait? contre les virus, il est de la responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot?g?s par le secret professionnel. Si vous d?sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier ?lectronique ordinaire non s?curis?, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/de53ad84/attachment.html From frank at carmickle.com Wed Dec 2 09:58:54 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 12:58:54 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B16A3F6.6000702@greatiam.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> Message-ID: <20091202175853.GW31924@base.carmickle.com> On Wed, Dec 02, Otis wrote: Snip... > Thanks. > > I would like all extensions on say server A to be contactable by those > on server B and vice versa. The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Mod_dptools#Applications http://wiki.freeswitch.org/wiki/Sofia --FC From mctch at yahoo.com Wed Dec 2 09:58:52 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 2 Dec 2009 09:58:52 -0800 (PST) Subject: [Freeswitch-users] call barge in In-Reply-To: Message-ID: <619084.32069.qm@web56408.mail.re3.yahoo.com> 1. Call recording: All (or selected) calls to the secretary must be recorded. Just requires an addition to the dialplan.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes ?emergency? (for example, an extension with emergency prefix) call to one of thesesubscribers -> Secretary barges in the established call (conference). In FreeSWITCH it is called eavesdrop and its in the default configuration in the dialplan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdropTo use it from the default entry in the dialplan you can dial 88 followed by the extension number. If you want to limit it to a set of extensions just add an additional condition. 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. It might be possible to do it with just the xml dialplan. However it could definitely be accomplished within a short time using a dialplan entry and a lua, perl or javascript. Best Regards, Mark J Crane http://www.fusionpbx.com (Open source graphical interface for FreeSWITCH) --- On Wed, 12/2/09, Artem Shiyanov wrote: From: Artem Shiyanov Subject: Re: [Freeswitch-users] call barge in To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 2, 2009, 10:21 AM 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Here is a bunch of code, search there ''barge" Artem On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote: Hi all, ? I?m evaluating FS for our organization. I must fulfill the following requirements: 1. Call recording: All (or selected) calls to the secretary must be recorded. 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes ?emergency? (for example, an extension with emergency prefix) call to one of these subscribers -> Secretary barges in the established call (conference). 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. ? Can anybody please advise if this is possible with FS? If yes, is it just a configuration task, or additional programming will be needed? ? Thanks in advance, Nikolay. ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/70d2ee0a/attachment-0001.html From msc at freeswitch.org Wed Dec 2 10:02:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:02:26 -0800 Subject: [Freeswitch-users] call barge in In-Reply-To: References: <20091202083449.271C2116CD@mail.nstel.ru> Message-ID: <87f2f3b90912021002p3db48213l9dc666774fa9ab25@mail.gmail.com> On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov wrote: > 1 - config > 2 - I've done this with programming > 3 - suppose programming would be needed > > Just to clarify, when you say "programming" there are different levels of involvement. For example, you can do programming in C which is pretty in depth, but that's probably not what is required. Most likely this all can be done with dialplan configuration and some simple Lua/Perl/JavaScript scripts. (We support many scripting languages.) I recommend that you install FreeSWITCH on a test server and connect a few phones. Start with the default configuration and make sure that you have it working properly and go from there. Also, we have an IRC channel on irc.freenode.net where you can come and discuss things realtime. Lastly, we have a weekly conference call where you can ask community members and developers your questions: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call I recommend that you use the latest SVN trunk as we are really close to 1.0.5. If you're on a Linux box you can do the quick install process mentioned here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Dive in and have fun! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/a919b7fd/attachment.html From frank at carmickle.com Wed Dec 2 10:05:48 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 13:05:48 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <20091202180548.GX31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : > > > > But still (and I don't even know if I'm using it the right way), I would > expect that when the voicemail greeting starts, I could press * on the > phone to call the vmain_unregistered_user extension in my dialplan, but > that never happens. > What I want, is when someone dials in (this is usefull only from outside > so via FXO line) his extension (or any other, it's not important) that > when > he reach the voicemail to leave a message, override the "leave a message" > voicemail and enter the "check my messages" voicemail (with an > authentication step). > > I'm not sure this is clear. Totally clear. My last example of extension vmain should have the goodies you need to have it work correctly. I have not tested changing to something different. Maybe this is broken? Try setting it back to the default and making your changes in the dialplan like I did. I can tell you that it is working as of svn 15396. --FC From msc at freeswitch.org Wed Dec 2 10:08:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:08:00 -0800 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <87f2f3b90912021008k1502b33dm434a6b19bf1f8792@mail.gmail.com> On Wed, Dec 2, 2009 at 9:43 AM, Fran?ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : > > > > But still (and I don't even know if I'm using it the right way), I would > expect that when the voicemail greeting starts, I could press * on the > phone to call the vmain_unregistered_user extension in my dialplan, but > that never happens. > What I want, is when someone dials in (this is usefull only from outside > so via FXO line) his extension (or any other, it's not important) that > when > he reach the voicemail to leave a message, override the "leave a message" > voicemail and enter the "check my messages" voicemail (with an > authentication step). > > I think you want to press zero: Of course, you can change that value to something else if you prefer the zero to be a transfer-to-operator kind of function. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/aa2211db/attachment.html From anthony.minessale at gmail.com Wed Dec 2 10:13:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:13:55 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> Message-ID: <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps. On Wed, Dec 2, 2009 at 10:35 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > As always, you are correct. > > The scenario now is: > > - If the caller places the callee on hold, the callee will get hold music > - If the callee places the caller on hold, the caller will not get hold > music > > I've uploaded a fresh pastebin here: > > http://pastebin.freeswitch.org/11356 > > On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale > wrote: > > results cant possibly be the same > > there is not even any broadcast involved in uuid_transfer ? > > > > you need to attach a console trace with debug log up > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/545cd94a/attachment.html From msc at freeswitch.org Wed Dec 2 10:13:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:13:59 -0800 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202175853.GW31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> Message-ID: <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > Snip... > > > Thanks. > > > > I would like all extensions on say server A to be contactable by those > > on server B and vice versa. > > The example I gave you should get you started. Let us know how you get > along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > Remember, too, that gateways are useful for doing auth/reg so having a gateway on each box that registers to the other box is pretty handy. If you run into any trouble trying to set it up you can ask here or join us in #freeswitch on irc.freenode.net. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/2b7e6c32/attachment.html From frank at carmickle.com Wed Dec 2 10:15:28 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 13:15:28 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <20091202181527.GY31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : Also, is there a functional requirement for two different extensions to call vmain? --FC From anthony.minessale at gmail.com Wed Dec 2 10:18:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:18:48 -0600 Subject: [Freeswitch-users] Dictation System In-Reply-To: <1259766025.1980.11.camel@dlaplap> References: <1259766025.1980.11.camel@dlaplap> Message-ID: <191c3a030912021018h78776ec0w51f18df795b78c1b@mail.gmail.com> Yes, I'm familiar with that application, check the src code for the author =p There has not been much of a demand for such an application but it's of course entirely possible to develop one. On Wed, Dec 2, 2009 at 9:00 AM, David Laperle wrote: > Hi Freeswitch users, > > i'm new into the PBX world. I just installed FreeSwitch and made work > great, but one of my goal with the PBX system is to use it as a dictation > system. We were using Callweaver, and there's a Dictation module for CW and > one for Asterisk, but i can't find one for FreeSwitch so far. Is there > anything in the trunk that i could find or any work in progress? > > I'm willing to develop a bit to help the work in progress, i have the > programming knowledge but not the VOIP/PBX knowledge, so a work in progress > could be enough for me to start with and complete the work! > > If any of you have an idea or a hint for me i would be very grateful! > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > *Robinson Sheppard Shapiro *s.e.n.c.r.l/LLP > Avocats / Barristers & Solicitors > 4600 - 800 Place Victoria > Montr?al Qc H4Z 1H6 > T (514) 878-2631 F (514) 878-1865 > www.rsslex.com et/and www.rsscanadaimmigration.com > > > > > * > ------------------------------ > **http://www.rsslex.com** * > > *AVIS:* Ce courriel privil?gi? et confidentiel est destin? ? la seule > personne ou entit? ? laquelle il est adress?. Pour toute autre personne, > toute action prise en rapport ? ce courriel ainsi que toute lecture, > reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble > de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? > recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. > Bien que ce courriel ait ?t? trait? contre les virus, il est de la > responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos > communications avec vous peuvent contenir des renseignements confidentiels > ou prot?g?s par le secret professionnel. Si vous d?sirez que nous > communiquions avec vous par un autre moyen de transmission que le courrier > ?lectronique ordinaire non s?curis?, veuillez nous en aviser. > > *NOTICE:* This privileged and confidential email is intended only for the > individual or entity to whom it is addressed. With regard to all others, any > action related with this email as well as any reading, reproduction, > transmission and/or dissemination in whole or in part of the information > included in this email is prohibited. If you are not the addressee, > immediately return the email to sender prior to destroying all copies. Even > if this email is believed to be free from any virus, it is the > responsibility of the recipient to make sure that it is virus exempt. Our > communications to you may contain confidential information or information > protected under solicitor-client privilege. Please advise if you wish us to > use a mode of communication other than regular, unsecured e-mail in our > communications with you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/7f0aaee3/attachment-0001.html From anthony.minessale at gmail.com Wed Dec 2 10:21:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:21:54 -0600 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Message-ID: <191c3a030912021021n23fa5981q8b6398587dfcb739@mail.gmail.com> bind to the transfer app so that it transfers the call to the vm extension that way the current application is always interrupted and replaced. The special "inline" dialplan lets you transfer calls right to an application use "inline" as the dp name and voicemail: as the extension On Wed, Dec 2, 2009 at 4:57 AM, Fran?ois Legal wrote: > Hello, > > > > I created an extension in my dialplan so that when an incoming call > arrives, it rings a group of lines and then fallback to the voicemail if no > line is answered. > > I wanted then that when voicemail starts, the calling party could dial some > numbers to fetch the voicemail. I used bind_meta_app for this. My problem > is, when using bind_meta_app, the voicemail continues, and I sometimes > experience freeswitch hanging after the call is over, depending on when the > bind_meta_app is activated. > > How can I make freeswitch terminate the first voicemail instance when > activating the bind_meta_app. > > > > Here's my extension : > > > > > > > > > > > > > > > Thanks > > > > Fran?ois > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0c04d795/attachment.html From msc at freeswitch.org Wed Dec 2 10:23:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:23:01 -0800 Subject: [Freeswitch-users] Dictation System In-Reply-To: <1259766025.1980.11.camel@dlaplap> References: <1259766025.1980.11.camel@dlaplap> Message-ID: <87f2f3b90912021023x138c3365t908ccd9cd877f66b@mail.gmail.com> This seems like an interesting niche project. I think that if you have programming skills then the community can provide the PBX/VoIP knowledge to help you get over the hump. I would recommend that you write up a document describing all the features that this module would need to provide. Reply to this thread when you have that ready. In the meantime I'll ask the community now: is there anyone else interested in this functionality that would like to help David get it off the ground? David, if you haven't already done so I recommend joining the freeswitch-dev mailing list and hopping on IRC in #freeswitch and #freeswitch-dev over on irc.freenode.net. Don't forget that we also have a weekly FreeSWITCH conference call each Friday so you can always hop on and discuss your ideas with some of the devs and other community members: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call Welcome to the community! -Michael (IRC: mercutioviz) On Wed, Dec 2, 2009 at 7:00 AM, David Laperle wrote: > Hi Freeswitch users, > > i'm new into the PBX world. I just installed FreeSwitch and made work > great, but one of my goal with the PBX system is to use it as a dictation > system. We were using Callweaver, and there's a Dictation module for CW and > one for Asterisk, but i can't find one for FreeSwitch so far. Is there > anything in the trunk that i could find or any work in progress? > > I'm willing to develop a bit to help the work in progress, i have the > programming knowledge but not the VOIP/PBX knowledge, so a work in progress > could be enough for me to start with and complete the work! > > If any of you have an idea or a hint for me i would be very grateful! > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/50c00df2/attachment.html From anthony.minessale at gmail.com Wed Dec 2 10:23:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:23:11 -0600 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> Message-ID: <191c3a030912021023y3663b48dgab739ad2e7cb95c@mail.gmail.com> you may have nat problems since I do not see many people reporting anything like this. On Wed, Dec 2, 2009 at 3:54 AM, Jonathan Barou wrote: > Any ideas ? > > Thanks > > 2009/11/30 Jonathan Barou > > My version is FreeSWITCH Version 1.0.trunk (15691M) >> >> http://jira.freeswitch.org/browse/FSBUILD-213 >> >> Thanks you. >> >> 2009/11/27 Anthony Minessale >> >> by latest do you mean SVN trunk? >>> >>> Can you issue the command "sofia profile internal siptrace on" before >>> capturing your trace and post the results >>> to http://pastebin.freeswitch.org or open a jira >>> http://jira.freeswitch.org on the issue and attach the log after you >>> create the issue ticket, don't include it in the mailing list. >>> >>> >>> On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: >>> >>>> Hi everybody, >>>> >>>> I'm actually using the lastest version of Freeswitch, I have a problem. >>>> I have a trunk SIP with my PABX. >>>> >>>> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >>>> 2. one Alcatel IpTouch 4028 with number 987 >>>> (on PABX) >>>> 3. one Siemens Gigaset A580 IP with number >>>> 8401 (on Freeswitch) >>>> >>>> >>>> *The first test* is to say to the phone 2 to transfer all the call to >>>> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >>>> phone 3 is ringing and it's work. I have that in the log : >>>> >>>> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >>>> early media >>>> >>>> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>>> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> >>>> 10.33.69.246 port 32000 codec: 8 ms: 90 >>>> >>>> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer >>>> [soft] 720 bytes per 90ms >>>> >>>> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >>>> >>>> v=0 >>>> >>>> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >>>> >>>> s=FreeSWITCH >>>> >>>> c=IN IP4 10.33.169.92 >>>> >>>> t=0 0 >>>> >>>> m=audio 23054 RTP/AVP 8 106 >>>> >>>> a=rtpmap:8 PCMA/8000 >>>> >>>> a=rtpmap:106 telephone-event/8000 >>>> >>>> a=fmtp:106 0-16 >>>> >>>> a=silenceSupp:off - - - - >>>> >>>> a=ptime:90 >>>> >>>> a=sendrecv >>>> >>>> >>>> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >>>> sofia/internal/368 at 10.33.69.246! >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>>> 368 at 10.33.69.246 skipping state [early][183] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >>>> sofia/internal/368 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>>> Activation Success L16 at 8000hz 1 channel 90ms >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >>>> Ringback Tone [%(2000,4000,440.0,480.0)] >>>> >>>> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >>>> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>>> >>>> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >>>> confirmed. >>>> >>>> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >>>> Buffer at 1440 bytes vs 81 >>>> >>>> >>>> >>>> *The Second Tes*t is to say to the phone 1 to transfer all the call to >>>> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >>>> one time and after it hangup. I have that in the log : >>>> >>>> >>>> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >>>> early media >>>> >>>> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>>> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> >>>> 10.33.69.144 port 32000 codec: 8 ms: 90 >>>> >>>> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer >>>> [soft] 720 bytes per 90ms >>>> >>>> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >>>> >>>> v=0 >>>> >>>> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >>>> >>>> s=FreeSWITCH >>>> >>>> c=IN IP4 10.33.169.92 >>>> >>>> t=0 0 >>>> >>>> m=audio 27732 RTP/AVP 8 106 >>>> >>>> a=rtpmap:8 PCMA/8000 >>>> >>>> a=rtpmap:106 telephone-event/8000 >>>> >>>> a=fmtp:106 0-16 >>>> >>>> a=silenceSupp:off - - - - >>>> >>>> a=ptime:90 >>>> >>>> a=sendrecv >>>> >>>> >>>> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >>>> sofia/internal/987 at 10.33.69.246! >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>>> 987 at 10.33.69.246 skipping state [early][183] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>>> Activation Success L16 at 8000hz 1 channel 90ms >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >>>> Ringback Tone [%(2000,4000,440.0,480.0)] >>>> >>>> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >>>> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >>>> 987 at 10.33.69.246 entering state [terminated][487] >>>> >>>> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >>>> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >>>> sofia/internal/987 at 10.33.69.246 [KILL] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 >>>> thread mismatch skipping state handler. >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >>>> original codec. >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >>>> [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >>>> other leg >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 hanging up, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING >>>> -> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on >>>> external entities >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 >>>> thread mismatch skipping state handler. >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session >>>> 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>>> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot >>>> create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >>>> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >>>> Cause: ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/987 at 10.33.69.246) State HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>>> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>>> sofia/internal/987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/987 at 10.33.69.246) State REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> >>>> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >>>> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session >>>> 47 (sofia/internal/987 at 10.33.69.246) Ended >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>>> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/987 at 10.33.69.246) State DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>>> 987 at 10.33.69.246 SOFIA DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/internal/987 at 10.33.69.246 Standard DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >>>> >>>> Finally when I tried to call the phone 3 with the phone 1 it's working, >>>> and not when I want to call the phone 3 with the phone 2, like just before, >>>> it's ringing just one time and hangup. >>>> >>>> >>>> Thanks you. >>>> >>>> >>>> Best Regards >>>> >>>> -- >>>> John >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Jonathan BAROU >> Groupe SQLI - CRCI >> >> 0472405368 >> jbarou at sqli.com >> >> 1, place Verrazzano >> 69258 LYON CEDEX 09 >> >> > > > -- > Jonathan BAROU > Groupe SQLI - CRCI > > 0472405368 > jbarou at sqli.com > 1, place Verrazzano > 69258 LYON CEDEX 09 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0a39440f/attachment-0001.html From erandr-junk at usa.net Wed Dec 2 10:23:51 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 10:23:51 -0800 (PST) Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> References: <26610094.post@talk.nabble.com> <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> Message-ID: <26613841.post@talk.nabble.com> I need a way to start a call from the PHP script to the originating number, tell the party on that number to hold on, start another call to destination number, and bridge everything together. On both legs I need to pass custom caller ID. I can of course open direct connections to VOIP gateways right from PHP, but I want to reuse existing routing rules in the dial plan, hence I want to know what's the best way of making originate go through a specific context of the dial plan. As for the number of calls per second, it's going to be only occasionally used. mercutioviz wrote: > > On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: > >> >> What would be the best way of making originate() run call through a dial >> plan >> (compared to directly going to a specified VOIP gateway). Would it be >> loopbacks, i.e. smth like this? >> >> /opt/freeswitch/bin/fs_cli -x "originate >> >> {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML >> '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js >> zzzzzzzzzz)'" >> >> The idea of this is that originate() sets up the first call, then >> webcall.js >> plays back a WAV, and bridges the first call with the second one (also >> set >> up via loopback). >> >> > Could you describe the problem that you're trying to solve? That would > make > it easier to know if what you've come up with is the best solution. How > many > calls per second were you wanting to generate with this setup? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26613841.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Wed Dec 2 10:32:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:32:21 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> References: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> Message-ID: <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> idle is a 4 letter word to a realtime application. The core keeps a single high-priority thread to keep 1ms timing and expands that broadcasting to hundreds or thousand of threads who need accurate timing. Your choppy audio is caused by linksys lying about the packet len that it's using and we set our timer to the wrong speed. On Tue, Dec 1, 2009 at 9:19 PM, wrote: > Wow... Thinking about this timer setting and about how it converted > send()/recv() from non-blocking to blocking, I straced freeswitch when it > was > supposed to be idle. It never pauses! It keeps going in and out of select() > every millisecond! Why?? > > ------ Original Message ------ > Received: Tue, 01 Dec 2009 08:31:46 PM EST > From: erandr-junk at usa.net > To: > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. > Just > > installing SVN trunk didn't fix it either, but setting that option > afterwards > > surely did the trick. > > > > One thing I've noticed while staring at the console is that it *looks > like* > > that w/o the new setting the stuttering happens when FS either > re-registers > > itself with the provider or one of the SPA's port re-registers with FS. > > > > ------ Original Message ------ > > Received: Tue, 01 Dec 2009 05:33:26 PM EST > > From: Anthony Minessale > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > > > linksys has had a bug for eons that can be fixed by setting the ptime > (or > > > rtp packet size in their terms) > > > in it's firmware to .20 instead of .30 > > > > > > Asterisk does not use async RTP like we do so it's never a problem > > > you can disable the timer by setting the channel var > rtp_timer_name=none > or > > > sofia param rtp-timer-name to none in the sofia profile. > > > > > > You should also test this on latest SVN trunk or wait for pre8 > > > > > > > > > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > > > > > > > I should also add, after browsing through some topics here, that my > SIP > > > > provider sends 172-byte RTP frames, which is in accordance with > ptime:20 > > > > that it gives to FreeSWITCH. > > > > > > > > > > > > eaf wrote: > > > > > > > > > > Hi, > > > > > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the > way > > > > how > > > > > it can be programmed), but ran into one issue with sound quality > that > I > > > > > just cannot workaround by myself. I would describe the sound > problem > as > > > > > being "choppy". From time to time small portions of the other > party's > > > > > voice are dropped, so the voice kind of stutters. This is not too > bad, > > > > but > > > > > is really noticeable, happens in every call and I don't experience > the > > > > > same with Asterisk running on the same box. I attached two files: > > > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > > > directly exposed to internet, and then dial a toll-free via > FutureNine > > (a > > > > > SIP provider). > > > > > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else > with > > > > this > > > > > provider. Only PCMU. Tried to match ptime of provider (30) with > ptime > > of > > > > > the SPA, didn't get any improvement. Tried turning off recording, > no > > > > > change either. > > > > > > > > > > What puzzles me is that even with greedy codec negotiations and > with > > PCMU > > > > > on both sides of FreeSWITCH, it's still saying that > > > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of > freeswitch.log > > > > to > > > > > illustrate. > > > > > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode > > LX800 > > > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope > that > > > > > it's not a performance issue. > > > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wavfreeswitch.wav > > > > > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.logfreeswitch.log > > > > > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, > and > > > > > consistently show no glitches with Asterisk. > > > > > > > > > > > > > > > > > > -- > > > > View this message in context: > > > > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com< > MSN%3Aanthony_minessale at hotmail.com > > > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org< > sip%3A888 at conference.freeswitch.org > > > > > iax:guest at conference.freeswitch.org/888 > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/a320b89e/attachment.html From davis.erwin at gmail.com Wed Dec 2 10:41:56 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 13:41:56 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match Message-ID: Hi, I got a weird issue when I dialed an extension and listen to a recorded voice mail greeting message. After playing a couple of time of the greeting, the FS printed the warning of "sample rate not matching", then send the audio to a different remote RTP port. See the log below, 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate doesn't match 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec Activated 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less than minimum record length: 3, discarding it. 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ the original codec is wideband 16kHz Speex and the wireshark shows that the FS used the same codec. I used FS 1.04 in fedora 8. I have two questions here, (1) why does FS report "Sample rate doesn't match"? is it a bug or configuration issue? (2) Why does FS change the RTP port ? how to fix it? Thanks, Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/8d7045db/attachment-0001.html From shiyanov at gmail.com Wed Dec 2 11:08:17 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 2 Dec 2009 22:08:17 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: I'm back again with the same issue. Now it is became worse: it reproduces occasionally. [FS version is 1.04, test_load = 2 active calls] I've got 2 logs: successful and not. Here is a bad_case: 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute java(/usr/local/freeswitch/scripts/fs2agi.jar org.starpound.fs2agi.Translator ${agi_url}) Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run INFO: *************************************************** Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI application agi://localhost:4573/hello.agi?callId=929 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/2001! 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel [sofia/internal/2001] has been answered Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: java.lang.Exception: Internal FreeSwitch failure while streamming file, see FreeSwitch logs for details at org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) at org.starpound.fs2agi.Translator.run(Translator.java:56) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.freeswitch.Launcher.launch(Launcher.java:80) 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for details. 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 (sofia/external/6786081291 at 66.19.38.143) Ended 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 (sofia/internal/2001) Ended 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2001 [CS_DESTROY] Message "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: ..." is sent from my app upon the onHangup().` And here is a good_case: 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute java(/usr/local/freeswitch/scripts/fs2agi.jar org.starpound.fs2agi.Translator ${agi_url}) Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run INFO: *************************************************** Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for session 7c37369b-ffb2-4436-9288-a640047d0e5e 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI application agi://localhost:4573/hello.agi?callId=932 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/2001! 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel [sofia/internal/2001] has been answered Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: java.lang.Exception: Internal FreeSwitch failure while streamming file, see FreeSwitch logs for details at org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) at org.starpound.fs2agi.Translator.run(Translator.java:56) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.freeswitch.Launcher.launch(Launcher.java:80) 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for details. 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from 172.26.10.39:26402 to 91.190.120.190:26402 Suggestions? On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: > Anthony, > > >>As soon as you call uuid_bridge you are transferring both legs of the > call to bridge to each other. > >>This means your java app must exit so the channels can connect to each > other. > > I didn't know that. Now my java app is exiting upon the onHangup() call so > everything has become "ok". Thank you much. > I'll add note to the wiki about this issue. > > Artem > > > > > On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Your "annoying behaviour" is the exact behavior you should be getting >> considering what you told FS to do. >> >> As soon as you call uuid_bridge you are transferring both legs of the call >> to bridge to each other. >> This means your java app must exit so the channels can connect to each >> other. >> >> remember that you hangup hook can be called when the channel is >> transferred not only when it hangs up. >> you have to test which is happening based on the input to your callback. >> >> >> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >> >>> Hi there! >>> >>> I've got annoying FS behavior: >>> There are 2 channels executing the same Java application (application >>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>> channels are killed. Here is a log from FS console: >>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>> CS_HIBERNATE >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>> CS_EXECUTE -> CS_HIBERNATE >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> >>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>> switch_core_session.c:933 Send signal >>> sofia/internal/1001 at master.agent.starpoundtec >>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>> >>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>> >>> (FS version is 1.0.4) >>> >>> Any thoughts? >>> >>> >>> Artem >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/f08f1532/attachment.html From anthony.minessale at gmail.com Wed Dec 2 11:17:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:17:55 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> Message-ID: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: > So I did some tests and still I can not see CLIP on a phone connected to an > FXS port. Whether the call is bridged from SIP UA or from an incoming call > on FXO port does not change anything. Whether the parameter > enable-caller-id=true is present or not in openzap.conf.xml does not change > anything too. > > On that subject, sangoma support team says it must be freeswitch as this > feature is supported and has been tested working. > > > > However, the good point is that I did not experience cuts in my call > bridged from FXS to FXO with that new release. > > > > Fran?ois > > > > On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > upgrading always helps *something* not sure. but that is where we have to > start because we have changed that code alot. > > > On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: > >> Sure, I'll try that. I'm just building freeswitch-snapshot that I >> downloaded from files.freeswitch.org >> >> I also experience, when bridging a call from an FXS to FXO the call is cut >> after a random time (this does not appear when bridging SIP to FXO). Might >> this upgrade fix this problem also ? >> >> >> >> Fran?ois >> >> >> >> On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: >> >> can you test svn trunk or latest pre release of 1.0.5 >> >> >> On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: >> >>> Hello, >>> >>> >>> >>> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP >>> problems on the FXS ports. >>> >>> When I ring on FXS ports, the connected phone does not display >>> callerid/callerid-name. >>> >>> I tried turning the stuff of in openzap.conf.xml () but it did not help. >>> >>> >>> >>> As a side note, turning this on on the FXO ports drops the callerid >>> information on incoming calls. >>> >>> >>> >>> Running freeswitch 1.0.4 on linux 2.6.27. >>> >>> >>> >>> Fran?ois >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/65c5d3d2/attachment-0001.html From anthony.minessale at gmail.com Wed Dec 2 11:24:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:24:09 -0600 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> you should be working on SVN trunk if you are doing development, we are so far forward from 1.0.4 we can't do debugging very easily. I don't know all of the details of what you are trying to do but you are hitting some race conditions because of the async nature of the socket connection and the way you are using it. On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov wrote: > I'm back again with the same issue. > Now it is became worse: it reproduces occasionally. > [FS version is 1.04, test_load = 2 active calls] > > I've got 2 logs: successful and not. > Here is a bad_case: > > 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute > java(/usr/local/freeswitch/scripts/fs2agi.jar > org.starpound.fs2agi.Translator > ${agi_url}) > Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run > INFO: *************************************************** > Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run > INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for > session > 2898ad41-4ec1-4628-89fd-651a93a7221d > 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI application > agi://localhost:4573/hello.agi?callId=929 > 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream > handle! > > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] > 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/2001! > 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel > [sofia/internal/2001] has > been answered > Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session > 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: > java.lang.Exception: Internal FreeSwitch failure while streamming file, see > FreeSwitch logs for details > at > > org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) > at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) > at org.starpound.fs2agi.Translator.run(Translator.java:56) > at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) > at > > sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) > at > > sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) > at java.lang.reflect.Method.invoke(Method.java:597) > at org.freeswitch.Launcher.launch(Launcher.java:80) > 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup > sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application > agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for > details. > 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup > sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/external/6786081291 at 66.19.38.143) Ended > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 > (sofia/internal/2001) Ended > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/internal/2001 [CS_DESTROY] > > > > Message > "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session > 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: > ..." > is sent from my app upon the onHangup().` > > And here is a good_case: > > 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute > java(/usr/local/freeswitch/scripts/fs2agi.jar > org.starpound.fs2agi.Translator > ${agi_url}) > Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run > INFO: *************************************************** > Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run > INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for > session > 7c37369b-ffb2-4436-9288-a640047d0e5e > 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI application > agi://localhost:4573/hello.agi?callId=932 > 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream > handle! > > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] > 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/2001! > 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel > [sofia/internal/2001] has > been answered > Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session > 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: > java.lang.Exception: Internal FreeSwitch failure while streamming file, see > FreeSwitch logs for details > at > > org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) > at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) > at org.starpound.fs2agi.Translator.run(Translator.java:56) > at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) > at > > sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) > at > > sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) > at java.lang.reflect.Method.invoke(Method.java:597) > at org.freeswitch.Launcher.launch(Launcher.java:80) > 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application > agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for > details. > 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from > 172.26.10.39:26402 to 91.190.120.190:26402 > > > > Suggestions? > > > > > > > > > > > > On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: > >> Anthony, >> >> >>As soon as you call uuid_bridge you are transferring both legs of the >> call to bridge to each other. >> >>This means your java app must exit so the channels can connect to each >> other. >> >> I didn't know that. Now my java app is exiting upon the onHangup() call so >> everything has become "ok". Thank you much. >> I'll add note to the wiki about this issue. >> >> Artem >> >> >> >> >> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Your "annoying behaviour" is the exact behavior you should be getting >>> considering what you told FS to do. >>> >>> As soon as you call uuid_bridge you are transferring both legs of the >>> call to bridge to each other. >>> This means your java app must exit so the channels can connect to each >>> other. >>> >>> remember that you hangup hook can be called when the channel is >>> transferred not only when it hangs up. >>> you have to test which is happening based on the input to your callback. >>> >>> >>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >>> >>>> Hi there! >>>> >>>> I've got annoying FS behavior: >>>> There are 2 channels executing the same Java application (application >>>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>>> channels are killed. Here is a log from FS console: >>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>>> CS_HIBERNATE >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>> called >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>>> playing file >>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>>> playing file >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >>>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>>> CS_EXECUTE -> CS_HIBERNATE >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>> called >>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>> >>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>>> switch_core_session.c:933 Send signal >>>> sofia/internal/1001 at master.agent.starpoundtec >>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >>>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>>> >>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >>>> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>>> >>>> (FS version is 1.0.4) >>>> >>>> Any thoughts? >>>> >>>> >>>> Artem >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0891a714/attachment.html From errotan at gmail.com Wed Dec 2 11:38:40 2009 From: errotan at gmail.com (=?utf-8?q?Pusk=C3=A1s_Zsolt?=) Date: Wed, 2 Dec 2009 20:38:40 +0100 Subject: [Freeswitch-users] CDR records In-Reply-To: <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> Message-ID: <200912022038.41102.errotan@gmail.com> 2009. december 2. 18.35.06 Michael Collins d?tummal ezt ?rta: > 2009/12/2 Jo?o Mesquita > > > What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid > > confusions. > > > > JM > > > > Thanks for catching my typo! :) > > -MC > Thanks for the hint I loaded mod_xml_cdr and now understand why there are 3 cdr records. I love mod_xml_cdr :) Thx! From anthony.minessale at gmail.com Wed Dec 2 11:46:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:46:31 -0600 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: Message-ID: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> you must only have 8k sounds so the resample is when it's playing files try make hd-sounds-install to install 16k sounds too On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: > Hi, I got a weird issue when I dialed an extension and listen to a recorded > voice mail greeting message. > After playing a couple of time of the greeting, the FS printed the warning > of "sample rate not matching", then > send the audio to a different remote RTP port. See the log below, > > > 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message.wav] (en:en) > 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate > doesn't match > 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec > Activated > 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from > xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 > 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original > codec. > 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less > than minimum record length: 3, discarding it. > 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-too-small.wav] (en:en) > 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ > > > the original codec is wideband 16kHz Speex and the wireshark shows that the > FS used the same codec. I used FS 1.04 in fedora 8. > I have two questions here, > (1) why does FS report "Sample rate doesn't match"? is it a bug or > configuration issue? > (2) Why does FS change the RTP port ? how to fix it? > > Thanks, > > Regards, > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/1b5e18de/attachment.html From msc at freeswitch.org Wed Dec 2 12:06:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 12:06:27 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: <200912022038.41102.errotan@gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> <200912022038.41102.errotan@gmail.com> Message-ID: <87f2f3b90912021206x1349a3belca432fa22ca961bf@mail.gmail.com> > I love mod_xml_cdr :) > > My sentiments as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/5bb524c4/attachment.html From davis.erwin at gmail.com Wed Dec 2 12:08:37 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 15:08:37 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: Hi, Anthony, Thanks for your reply. When I type the command below, I got the error, Unknown target hd-sound-install make[1]: *** [hd-sound-install] Error 1 make: *** [hd-sound-install] Error 2 I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should explain why at first FS played in right sample rate. But after playing serveral time, FS complained about sample rate not matching. Any clue? Thanks, On 12/2/09, Anthony Minessale wrote: > > you must only have 8k sounds so the resample is when it's playing files > > try make hd-sounds-install to install 16k sounds too > > > > > > On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: > >> Hi, I got a weird issue when I dialed an extension and listen to a >> recorded voice mail greeting message. >> After playing a couple of time of the greeting, the FS printed the warning >> of "sample rate not matching", then >> send the audio to a different remote RTP port. See the log below, >> >> >> 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language >> specified - Using [en] >> 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle >> play-file:[voicemail/vm-record_message.wav] (en:en) >> 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate >> doesn't match >> 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec >> Activated >> 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port >> from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 >> 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore >> original codec. >> 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less >> than minimum record length: 3, discarding it. >> 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language >> specified - Using [en] >> 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle >> play-file:[voicemail/vm-too-small.wav] (en:en) >> 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ >> >> >> the original codec is wideband 16kHz Speex and the wireshark shows that >> the FS used the same codec. I used FS 1.04 in fedora 8. >> I have two questions here, >> (1) why does FS report "Sample rate doesn't match"? is it a bug or >> configuration issue? >> (2) Why does FS change the RTP port ? how to fix it? >> >> Thanks, >> >> Regards, >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/9f8ee643/attachment.html From kristian.kielhofner at gmail.com Wed Dec 2 12:16:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 2 Dec 2009 15:16:38 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> Message-ID: <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> Tony, Thanks for that but now it appears that the call just gets hung up on when the caller takes the callee off hold. Debug here: http://pastebin.freeswitch.org/11359 Thanks again! On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale wrote: > I decided to just change the code so its more elegant to handle recursive > broadcasting so you can try again and see if that helps. > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Dec 2 12:21:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 14:21:14 -0600 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: <191c3a030912021221r56ed768ei16dfe448d51a4a23@mail.gmail.com> that was make hd-sounds-install sorrry you should also update to SVN trunk because based on the line number in your log its clear you are using a much older version of FS On Wed, Dec 2, 2009 at 2:08 PM, Erwin Davis wrote: > Hi, Anthony, > > Thanks for your reply. > > When I type the command below, I got the error, > Unknown target hd-sound-install > make[1]: *** [hd-sound-install] Error 1 > make: *** [hd-sound-install] Error 2 > > I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, > there are directories, 8000, 16000, 32000, 48000 for recorded voicemail > greetings. It should explain why at first FS played in right sample rate. > But after playing serveral time, FS complained about sample rate not > matching. Any clue? Thanks, > > > > > > On 12/2/09, Anthony Minessale wrote: >> >> you must only have 8k sounds so the resample is when it's playing files >> >> try make hd-sounds-install to install 16k sounds too >> >> >> >> >> >> On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: >> >>> Hi, I got a weird issue when I dialed an extension and listen to a >>> recorded voice mail greeting message. >>> After playing a couple of time of the greeting, the FS printed the >>> warning of "sample rate not matching", then >>> send the audio to a different remote RTP port. See the log below, >>> >>> >>> 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing >>> file >>> 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language >>> specified - Using [en] >>> 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle >>> play-file:[voicemail/vm-record_message.wav] (en:en) >>> 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate >>> doesn't match >>> 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec >>> Activated >>> 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port >>> from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 >>> 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore >>> original codec. >>> 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less >>> than minimum record length: 3, discarding it. >>> 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language >>> specified - Using [en] >>> 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle >>> play-file:[voicemail/vm-too-small.wav] (en:en) >>> 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ >>> >>> >>> the original codec is wideband 16kHz Speex and the wireshark shows that >>> the FS used the same codec. I used FS 1.04 in fedora 8. >>> I have two questions here, >>> (1) why does FS report "Sample rate doesn't match"? is it a bug or >>> configuration issue? >>> (2) Why does FS change the RTP port ? how to fix it? >>> >>> Thanks, >>> >>> Regards, >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/31bfbd81/attachment.html From gmartinr82 at gmail.com Wed Dec 2 10:04:58 2009 From: gmartinr82 at gmail.com (Martin Rodriguez) Date: Wed, 2 Dec 2009 15:04:58 -0300 Subject: [Freeswitch-users] First steps in FreeSWITCH Message-ID: Hi list; I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and 10 years in VoIP (Cisco). I need a reference guide to start working with FreeSWITCH. I download the official documentation, it would need some other configuration examples and dialplan sip device for calls. Martin Rodriguez From msc at freeswitch.org Wed Dec 2 12:57:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 12:57:18 -0800 Subject: [Freeswitch-users] First steps in FreeSWITCH In-Reply-To: References: Message-ID: <87f2f3b90912021257y6b2bea48lfb31a15ce151fcf@mail.gmail.com> On Wed, Dec 2, 2009 at 10:04 AM, Martin Rodriguez wrote: > Hi list; > > I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and > 10 years in VoIP (Cisco). I need a reference guide to start working > with > FreeSWITCH. I download the official documentation, it would need some > other configuration examples and dialplan sip device for calls. > > Martin Rodriguez > > Welcome to FreeSWITCH! We think you'll like it. Just remember that it's different than Cisco and Asterisk. (By "different" I mean "more powerful, better thought out, more flexible, and way, WAY cooler" :) ) First off you might want to peruse the Rosetta Stone: http://wiki.freeswitch.org/wiki/Rosetta_stone It's a place where you can get a frame of reference. If you've worked with Asterisk for six years then you've got lots of knowledge, and probably a few battle scars :) and that document helps you put things into perspective a bit. If you want a gentle intro to installing and setting up FS then check out this article: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT Note: the article mentions version 1.0.4 but we're really close to 1.0.5. We strongly recommend everyone, especially experienced users, to use the latest SVN trunk as it is almost always the most stable version of FS. Lastly, be sure to check out the getting started guide and the installation guide on the wiki. (www.freeswitch.org) The quick-and-dirty install guide is good for those who now Linux and just want to jump right in. -MC (IRC: mercutioviz) P.S. - we have a weekly FS community conf call and you're welcome to join: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/41ff6e43/attachment.html From msc at freeswitch.org Wed Dec 2 13:03:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 13:03:02 -0800 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: > Hi, Anthony, > > Thanks for your reply. > > When I type the command below, I got the error, > Unknown target hd-sound-install > make[1]: *** [hd-sound-install] Error 1 > make: *** [hd-sound-install] Error 2 > > I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, > there are directories, 8000, 16000, 32000, 48000 for recorded voicemail > greetings. It should explain why at first FS played in right sample rate. > But after playing serveral time, FS complained about sample rate not > matching. Any clue? Thanks, > > Erwin, As Tony said you've actually got a pretty old installation. If this is in production then I would recommend getting a sandbox machine, install trunk using the quick-and-dirty install, and then update the default config to you specific configuration. Test to make sure it works before you put it into production. :) Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run into any issues that require more real-time conversation. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/45cd76fa/attachment-0001.html From davis.erwin at gmail.com Wed Dec 2 13:15:17 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 16:15:17 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> Message-ID: Hi, Anthony and Mike, Thanks for your reply. The problem still exists even after I ran "make hd-sounds install". I will try the latest version from the SVN to see if the problem will go away. I will let you know. Thanks folks, Regards, On 12/2/09, Michael Collins wrote: > > > > On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: > >> Hi, Anthony, >> >> Thanks for your reply. >> >> When I type the command below, I got the error, >> Unknown target hd-sound-install >> make[1]: *** [hd-sound-install] Error 1 >> make: *** [hd-sound-install] Error 2 >> >> I found out that under >> /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, >> 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should >> explain why at first FS played in right sample rate. But after playing >> serveral time, FS complained about sample rate not matching. Any clue? >> Thanks, >> >> > Erwin, > > As Tony said you've actually got a pretty old installation. If this is in > production then I would recommend getting a sandbox machine, install trunk > using the quick-and-dirty install, and then update the default config to you > specific configuration. Test to make sure it works before you put it into > production. :) > > Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run > into any issues that require more real-time conversation. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/05b03e36/attachment.html From larclap at yahoo.com Wed Dec 2 14:07:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 14:07:38 -0800 Subject: [Freeswitch-users] Eavesdrop error? Message-ID: <036401ca739b$dc451340$94cf39c0$@com> I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars From abeka at greatiam.com Wed Dec 2 14:44:42 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 22:44:42 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202175853.GW31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> Message-ID: <4B16EDDA.5040908@greatiam.com> Thanks. Will let you know Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > Snip... > > >> Thanks. >> >> I would like all extensions on say server A to be contactable by those >> on server B and vice versa. >> > > The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > > From larclap at yahoo.com Wed Dec 2 15:22:20 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 15:22:20 -0800 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <036401ca739b$dc451340$94cf39c0$@com> References: <036401ca739b$dc451340$94cf39c0$@com> Message-ID: <03b101ca73a6$4c8db080$e5a91180$@com> Sorry, svn 15753 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Wednesday, December 02, 2009 2:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Eavesdrop error? I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Wed Dec 2 15:33:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 17:33:22 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> Message-ID: <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> I am not sure what you are sending over the socket but you have a queued hangup being processed on line 640 of your pastebin are you executing any commands with a ! character in it by any chance or executing the hangup app on purpose? On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Tony, > > Thanks for that but now it appears that the call just gets hung up > on when the caller takes the callee off hold. Debug here: > > http://pastebin.freeswitch.org/11359 > > Thanks again! > > On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale > wrote: > > I decided to just change the code so its more elegant to handle recursive > > broadcasting so you can try again and see if that helps. > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/1dad3d49/attachment.html From anthony.minessale at gmail.com Wed Dec 2 15:34:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 17:34:57 -0600 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03b101ca73a6$4c8db080$e5a91180$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> Message-ID: <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> it probably just means the uuid was not retrieved from the db when you called the eavesdrop exten which does the lookup on the uuid for the hash key based on what ext you hit to retrieve the most recent uuid that called that ext. On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > Sorry, svn 15753 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars > Zeb > Sent: Wednesday, December 02, 2009 2:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Eavesdrop error? > > I tried to use eavesdrop today and it did not work. The error message in > the > log is: > > [ERR] mod_dptools.c:334 Usage: [all | ] > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > incorrect? > > http://pastebin.freeswitch.org/11363 > > Thanks Lars > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/95347fd4/attachment-0001.html From erandr-junk at usa.net Wed Dec 2 16:31:31 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 16:31:31 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> Message-ID: <26619085.post@talk.nabble.com> Can I reduce resolution of that timer thread 10 times? I mean, I glanced through the code, and see that among others (are there others?) RTP and IVR set up their timers that are subsequently managed by this thread. RTP timers should be eliminated by that setting you've suggested. IVR timers are set at 20ms... So, if the thread is set to wake up every 10ms instead of 1ms it should be able to wake up those IVR timers just fine. Right? That's a cool design to have one dedicated thread that maintains accurate timing and then broadcasts via condition variables to hundreds of other threads events that they can register for. I'm sure it's one of the reasons why FS scales so much better than Asterisk. But for poor low-end setups that sit in the closet, eat only 6W of power and hardly ever run more than two calls at the same time, can I hack it somehow to be more UNIX-friendly? I.e. make it stuck in select() or recv() when there is nothing to do, call clock_gettime() right from the thread that wants and when it wants to know current time? Say, what if that thread is made to suspend on a condition variable in case if there are no timers registered in TIMER_MATRIX? Then, if some other thread comes up and adds its timer into the matrix, it could wake up the timer thread and enjoy accurate timing as needed, on demand? And in-between the calls, when there is no RTP or IVR, it will all go silent? I mean, sitting on a wait queue in the kernel is way better than go back and forth incrementing counters that nobody even needs at the moment? Anthony Minessale-2 wrote: > > idle is a 4 letter word to a realtime application. > > The core keeps a single high-priority thread to keep 1ms timing and > expands > that broadcasting > to hundreds or thousand of threads who need accurate timing. > > Your choppy audio is caused by linksys lying about the packet len that > it's > using and we set our timer > to the wrong speed. > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From timuckun at gmail.com Wed Dec 2 16:49:36 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 3 Dec 2009 13:49:36 +1300 Subject: [Freeswitch-users] HA questions. Message-ID: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to "lose" those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. From larclap at yahoo.com Wed Dec 2 18:19:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 18:19:58 -0800 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> Message-ID: <03c401ca73bf$1cea8600$56bf9200$@com> Is this reasonable given it was the only call in FreeSwitch at the time? How can this situation be corrected in the future? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, December 02, 2009 3:35 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Eavesdrop error? it probably just means the uuid was not retrieved from the db when you called the eavesdrop exten which does the lookup on the uuid for the hash key based on what ext you hit to retrieve the most recent uuid that called that ext. On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: Sorry, svn 15753 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Wednesday, December 02, 2009 2:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Eavesdrop error? I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From erandr-junk at usa.net Wed Dec 2 19:35:39 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 19:35:39 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26619085.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> Message-ID: <26620518.post@talk.nabble.com> Oh, looks like the timers are also used for streaming local data in read_stream_thread(). Due to this there is always one timer active with 20ms interval. But wait a sec, why is freeswitch periodically trying to stream /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav somewhere? Every minute or so? Did I misconfigure it? eaf wrote: > > Say, what if that thread is made to suspend on a condition variable in > case if there are no timers registered in TIMER_MATRIX? Then, if some > other thread comes up and adds its timer into the matrix, it could wake up > the timer thread and enjoy accurate timing as needed, on demand? And > in-between the calls, when there is no RTP or IVR, it will all go silent? > I mean, sitting on a wait queue in the kernel is way better than go back > and forth incrementing counters that nobody even needs at the moment? > > > Anthony Minessale-2 wrote: >> >> idle is a 4 letter word to a realtime application. >> >> The core keeps a single high-priority thread to keep 1ms timing and >> expands >> that broadcasting >> to hundreds or thousand of threads who need accurate timing. >> >> Your choppy audio is caused by linksys lying about the packet len that >> it's >> using and we set our timer >> to the wrong speed. >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620518.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at psrm.ca Wed Dec 2 19:35:59 2009 From: john at psrm.ca (John Lalande) Date: Wed, 2 Dec 2009 21:35:59 -0600 Subject: [Freeswitch-users] can't register Inphonex Message-ID: <005a01ca73c9$bc2dcf60$34896e20$@ca> I am new to FS having ditched Asterisk a few weeks ago. I have iptel registered but I cannot get Inphonex to work. I am using the settings from http://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no avail. The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout [408]." Is there some way to debug this? sofia status displays: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at 192.168.125.15:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG inphonex gateway sip:5285418 at sip.inphonex.com FAILED (retry: 28s) iptel gateway sip:jlalande at sip.iptel.org REGED internal profile sip:mod_sofia at 192.168.125.15:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) 192.168.125.15 alias internal ALIASED ============================================================================ ===================== -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/43b7aa50/attachment.html From erandr-junk at usa.net Wed Dec 2 19:47:42 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 19:47:42 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26620518.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <26620518.post@talk.nabble.com> Message-ID: <26620610.post@talk.nabble.com> OK, I'm slow. It's music-on-hold, and it's playing non-stop like that timer thread. Even when there are no calls. Why? eaf wrote: > > Oh, looks like the timers are also used for streaming local data in > read_stream_thread(). Due to this there is always one timer active with > 20ms interval. > > But wait a sec, why is freeswitch periodically trying to stream > /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav somewhere? > Every minute or so? Did I misconfigure it? > > > eaf wrote: >> >> Say, what if that thread is made to suspend on a condition variable in >> case if there are no timers registered in TIMER_MATRIX? Then, if some >> other thread comes up and adds its timer into the matrix, it could wake >> up the timer thread and enjoy accurate timing as needed, on demand? And >> in-between the calls, when there is no RTP or IVR, it will all go silent? >> I mean, sitting on a wait queue in the kernel is way better than go back >> and forth incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620610.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Dec 2 19:57:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Dec 2009 22:57:06 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26620518.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <26620518.post@talk.nabble.com> Message-ID: <3377EB5F-5022-4D72-B56C-61B3ED25A304@jerris.com> This is keeping track of a place in the music on hold so your hold music does not start back up at the same place every time. If you don't want to do this it is a module that you don't need to load and you can get your moh from any soundfile at your choice in configuration. Mike On Dec 2, 2009, at 10:35 PM, eaf wrote: > > Oh, looks like the timers are also used for streaming local data in > read_stream_thread(). Due to this there is always one timer active > with 20ms > interval. > > But wait a sec, why is freeswitch periodically trying to stream > /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav > somewhere? > Every minute or so? Did I misconfigure it? > > > eaf wrote: >> >> Say, what if that thread is made to suspend on a condition variable >> in >> case if there are no timers registered in TIMER_MATRIX? Then, if some >> other thread comes up and adds its timer into the matrix, it could >> wake up >> the timer thread and enjoy accurate timing as needed, on demand? And >> in-between the calls, when there is no RTP or IVR, it will all go >> silent? >> I mean, sitting on a wait queue in the kernel is way better than go >> back >> and forth incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len >>> that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620518.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Wed Dec 2 20:00:00 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Dec 2009 23:00:00 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26619085.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> Message-ID: <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> In short. No, you can not for many reasons. The milisecond tic is used throughout the code even when there is not any calls up. You can grep for switch_cond_next if you would like to see where but it is required to keep our global timestamp and for pacing the scheduler among other services that run all the time. Mike On Dec 2, 2009, at 7:31 PM, eaf wrote: > > Can I reduce resolution of that timer thread 10 times? I mean, I > glanced > through the code, and see that among others (are there others?) RTP > and IVR > set up their timers that are subsequently managed by this thread. > RTP timers > should be eliminated by that setting you've suggested. IVR timers > are set at > 20ms... So, if the thread is set to wake up every 10ms instead of > 1ms it > should be able to wake up those IVR timers just fine. Right? > > That's a cool design to have one dedicated thread that maintains > accurate > timing and then broadcasts via condition variables to hundreds of > other > threads events that they can register for. I'm sure it's one of the > reasons > why FS scales so much better than Asterisk. But for poor low-end > setups that > sit in the closet, eat only 6W of power and hardly ever run more > than two > calls at the same time, can I hack it somehow to be more UNIX- > friendly? I.e. > make it stuck in select() or recv() when there is nothing to do, call > clock_gettime() right from the thread that wants and when it wants > to know > current time? > > Say, what if that thread is made to suspend on a condition variable > in case > if there are no timers registered in TIMER_MATRIX? Then, if some other > thread comes up and adds its timer into the matrix, it could wake up > the > timer thread and enjoy accurate timing as needed, on demand? And in- > between > the calls, when there is no RTP or IVR, it will all go silent? I mean, > sitting on a wait queue in the kernel is way better than go back and > forth > incrementing counters that nobody even needs at the moment? > > > Anthony Minessale-2 wrote: >> >> idle is a 4 letter word to a realtime application. >> >> The core keeps a single high-priority thread to keep 1ms timing and >> expands >> that broadcasting >> to hundreds or thousand of threads who need accurate timing. >> >> Your choppy audio is caused by linksys lying about the packet len >> that >> it's >> using and we set our timer >> to the wrong speed. >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From erandr-junk at usa.net Wed Dec 2 20:58:39 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 20:58:39 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> Message-ID: <26621005.post@talk.nabble.com> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides that. Yeah, there is a global timestamp... It's easy to workaround that for RTP who calls switch_micro_time_now()... But if somebody accesses runtime.timestamp directly, it's gonna be tough to grep for that. If only this was C++... I'll play around. Never liked polling too much. Never could've guessed that polling could be so useful for scalability ;) My naive implementation would've pulled timestamp via system calls and would've done sleeping by passing exact interval to select() instead of syncing with a pacing thread. Which would be dead-quiet at idle time, but, of course, would stop scaling at some point due to excessive number of system calls. Thanks. Michael Jerris wrote: > > In short. No, you can not for many reasons. The milisecond tic is > used throughout the code even when there is not any calls up. You can > grep for switch_cond_next if you would like to see where but it is > required to keep our global timestamp and for pacing the scheduler > among other services that run all the time. > > Mike > > On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >> Can I reduce resolution of that timer thread 10 times? I mean, I >> glanced >> through the code, and see that among others (are there others?) RTP >> and IVR >> set up their timers that are subsequently managed by this thread. >> RTP timers >> should be eliminated by that setting you've suggested. IVR timers >> are set at >> 20ms... So, if the thread is set to wake up every 10ms instead of >> 1ms it >> should be able to wake up those IVR timers just fine. Right? >> >> That's a cool design to have one dedicated thread that maintains >> accurate >> timing and then broadcasts via condition variables to hundreds of >> other >> threads events that they can register for. I'm sure it's one of the >> reasons >> why FS scales so much better than Asterisk. But for poor low-end >> setups that >> sit in the closet, eat only 6W of power and hardly ever run more >> than two >> calls at the same time, can I hack it somehow to be more UNIX- >> friendly? I.e. >> make it stuck in select() or recv() when there is nothing to do, call >> clock_gettime() right from the thread that wants and when it wants >> to know >> current time? >> >> Say, what if that thread is made to suspend on a condition variable >> in case >> if there are no timers registered in TIMER_MATRIX? Then, if some other >> thread comes up and adds its timer into the matrix, it could wake up >> the >> timer thread and enjoy accurate timing as needed, on demand? And in- >> between >> the calls, when there is no RTP or IVR, it will all go silent? I mean, >> sitting on a wait queue in the kernel is way better than go back and >> forth >> incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len >>> that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26621005.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Wed Dec 2 21:11:59 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 07:11:59 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO Message-ID: Hello, I have Polycom phones which send only RFC-2833 (or inband which I dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco gateway has some bug and accepts only INFO. I did a few tests: - Some of the phones are on different profile than the Cisco. On their profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set 'dtmf-type=info' and Freeswitch did the translation. All works ok... - Some of the phones are on the same profile as the Cisco, so I must set dtmf-type to rfc2833; it works with internal applications (like voicemail) but does not work through the Cisco as it misinterprets the rfc2833 Is there a way to set some variable (or a parameter to the bridge application) to do the translation? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/7160286f/attachment.html From jingwei.yang at gmail.com Wed Dec 2 22:02:47 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:02:47 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c Message-ID: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function ???X11_errors_handler???: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ???skypiax_send_message???: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/d0dab081/attachment.html From mrene_lists at avgs.ca Wed Dec 2 22:09:53 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Dec 2009 01:09:53 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> Message-ID: <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > Hi Guys, > > I got a compilation error of skypiax_protocol.c with the latest > version r15764. > > Compiling skypiax_protocol.c... > cc1: warnings being treated as errors > skypiax_protocol.c: In function ???X11_errors_handler???: > skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c: In function ???skypiax_send_message???: > skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: > skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations > and code > make[5]: *** [skypiax_protocol.o] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_skypiax-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > I personally checked the file and it shouldn't be a merge problem. > Does anyone encounter this as well? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8a36213f/attachment.html From jingwei.yang at gmail.com Wed Dec 2 22:33:16 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:33:16 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> Message-ID: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > Consider it fixed. > Committed revision 15765. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > > Hi Guys, > > I got a compilation error of skypiax_protocol.c with the latest version > r15764. > > Compiling skypiax_protocol.c... > *cc1: warnings being treated as errors* > skypiax_protocol.c: In function ???X11_errors_handler???: > skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c: In function ???skypiax_send_message???: > skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: > skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and > code > make[5]: *** [skypiax_protocol.o] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_skypiax-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > I personally checked the file and it shouldn't be a merge problem. Does > anyone encounter this as well? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b632a9a9/attachment-0001.html From mrene_lists at avgs.ca Wed Dec 2 22:39:16 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Dec 2009 01:39:16 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Message-ID: <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG - > std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings - > Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden - > DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o > make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/ > tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/ > libtiff.a -ljpeg -lz -lm -lc > ./make_at_dictionary: error while loading shared libraries: > libjpeg.so.7: cannot open shared object file: No such file or > directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene > wrote: > Consider it fixed. > Committed revision 15765. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest >> version r15764. >> >> Compiling skypiax_protocol.c... >> cc1: warnings being treated as errors >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func? >> ??: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed >> declarations and code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. >> Does anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c91d9e51/attachment.html From jingwei.yang at gmail.com Wed Dec 2 22:43:30 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:43:30 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Message-ID: <13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com> Not sure whether this error is due to the lack of libjpeg. I just double checked, this library had been installed. Package libjpeg-6b-37.i386 already installed and latest version On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang wrote: > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc > ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: > cannot open shared object file: No such file or directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest version >> r15764. >> >> Compiling skypiax_protocol.c... >> *cc1: warnings being treated as errors* >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >> code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. Does >> anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/66b655b8/attachment.html From yehavi.bourvine at gmail.com Wed Dec 2 22:48:12 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 08:48:12 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Message-ID: BTW, I forgot to update: I changed the bridge parameters to use sofia_contact() and it solved the problem. I also fixed the presence problem I had before with sofia_contact() (added presence_id to the bridge command). Regards, __Yehavi: 2009/11/24 Yehavi Bourvine > Hello Anthony, > > Indeed I see the reference to this channel variable in the code, but when > trying to access it from the dial plan it is empty... I try to get the value > of ${sip_profile_name} and it is empty. > > Thanks! __Yehavi: > > 2009/11/23 Anthony Minessale > >> Let's just do this: >> >> r15629 or higher >> >> look for sip_profile_name >> >> >> >> >> On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: >> >>> Hi >>> We have more then one profile. To make a call I have to enter : bridge >>> sofia/profile/number at ip >>> The problem is when I use : "${use_profile}" I am getting the caller >>> profile, and I need the destination profile. >>> >>> How do I get this information? >>> >>> Thanks >>> >>> Eli >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b271714d/attachment-0001.html From jingwei.yang at gmail.com Wed Dec 2 22:49:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:49:35 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> Message-ID: <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > Hi, > > That one is on your side. If you changed/updated system libs it might be > worth doing another ./configure > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc > ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: > cannot open shared object file: No such file or directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest version >> r15764. >> >> Compiling skypiax_protocol.c... >> *cc1: warnings being treated as errors* >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >> code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. Does >> anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/74598504/attachment.html From devel at thom.fr.eu.org Thu Dec 3 00:00:55 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 09:00:55 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202181527.GY31924@base.carmickle.com> References: <20091202181527.GY31924@base.carmickle.com> Message-ID: <9b3453c2ea2580e6080cfb4dbd6e3d7e@thom.fr.eu.org> Well, I'm just starting to use freeswitch, so my approach is probably for from optimal. The point is I wanted that voicemail do not prompt for passwords when the caller is a sip registered user, but I also wanted the login requirement if the voicemail was called from some FXS port. That lead me to having : in my dialplan. Fran?ois On Wed, 2 Dec 2009 13:15:28 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> No, my voicemail extension (I have 2 actually) is called >> vmain_unregistered_user, so in voicemail.conf.xml I have : > > Also, is there a functional requirement for two different extensions to > call vmain? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jingwei.yang at gmail.com Thu Dec 3 00:24:04 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 16:24:04 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> Message-ID: <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ???command_search???: at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in this function) at_interpreter.c: In function ???at_interpreter???: at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > No, I didn't change or update the system libs. I just wanted to double > check whether my system has this libjpeg library. ./configure was definitely > executed before the source codes were rebuilt. > > Regards, > -Jingwei > > > On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > >> Hi, >> >> That one is on your side. If you changed/updated system libs it might be >> worth doing another ./configure >> >> Cheers, >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >> >> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >> However, I encounter another one. >> >> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> -lc >> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: >> cannot open shared object file: No such file or directory >> make[8]: *** [at_interpreter_dictionary.h] Error 127 >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> Do you have idea about this one? >> >> Thanks! >> >> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >> >>> Consider it fixed. >>> Committed revision 15765. >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>> >>> Hi Guys, >>> >>> I got a compilation error of skypiax_protocol.c with the latest version >>> r15764. >>> >>> Compiling skypiax_protocol.c... >>> *cc1: warnings being treated as errors* >>> skypiax_protocol.c: In function ???X11_errors_handler???: >>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ???skypiax_send_message???: >>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >>> code >>> make[5]: *** [skypiax_protocol.o] Error 1 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_skypiax-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> I personally checked the file and it shouldn't be a merge problem. Does >>> anyone encounter this as well? >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b6d1f51c/attachment-0001.html From oseslija at gmail.com Thu Dec 3 00:43:28 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 3 Dec 2009 09:43:28 +0100 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: Message-ID: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Bear in mind that FS will accept both 2833 and INFO in any profile on an inbound call. Param "dtmf-type" is valid only for outbound calls from the profile. Ognjen On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine wrote: > Hello, > > I have Polycom phones which send only RFC-2833 (or inband which I > dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco > gateway has some bug and accepts only INFO. > > I did a few tests: > > - Some of the phones are on different profile than the Cisco. On their > profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set > 'dtmf-type=info' and Freeswitch did the translation. All works ok... > - Some of the phones are on the same profile as the Cisco, so I must > set dtmf-type to rfc2833; it works with internal applications (like > voicemail) but does not work through the Cisco as it misinterprets the > rfc2833 > > > Is there a way to set some variable (or a parameter to the bridge > application) to do the translation? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/dbcf0a36/attachment.html From kond at nstel.ru Thu Dec 3 00:44:31 2009 From: kond at nstel.ru (Nikolay Kondratyev) Date: Thu, 3 Dec 2009 11:44:31 +0300 Subject: [Freeswitch-users] call barge in In-Reply-To: <87f2f3b90912021002p3db48213l9dc666774fa9ab25@mail.gmail.com> Message-ID: <20091203084431.DC45D11A9D@mail.nstel.ru> Michael, Mark, Artem, Thank you for your answers. I believe FS will suite our needs. I've installed dedicated virtual machine (Centos) for FS and going to play with it. Thanks and regards, Nikolay. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 02, 2009 9:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] call barge in On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov wrote: 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Just to clarify, when you say "programming" there are different levels of involvement. For example, you can do programming in C which is pretty in depth, but that's probably not what is required. Most likely this all can be done with dialplan configuration and some simple Lua/Perl/JavaScript scripts. (We support many scripting languages.) I recommend that you install FreeSWITCH on a test server and connect a few phones. Start with the default configuration and make sure that you have it working properly and go from there. Also, we have an IRC channel on irc.freenode.net where you can come and discuss things realtime. Lastly, we have a weekly conference call where you can ask community members and developers your questions: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call I recommend that you use the latest SVN trunk as we are really close to 1.0.5. If you're on a Linux box you can do the quick install process mentioned here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Dive in and have fun! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/af25ea7a/attachment.html From jingwei.yang at gmail.com Thu Dec 3 01:29:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 17:29:08 +0800 Subject: [Freeswitch-users] OpenZap issues with incoming and outgoing calls Message-ID: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> Hello All, I have a Digium TDM400P pci card with two FXO ports installed on my linux box. I've connected an external telephone line to the first FXO port. But I can't either make outgoing calls or receive incoming ones. Here are my setups, please let me know where goes wrong. * /etc/zaptel.conf* loadzone = sg defaultzone=sg fxsks=1,2 */usr/local/freeswitch/conf/zt.conf* remains unchanged [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 */usr/local/freeswitch/conf/openzap.conf* [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* I defined an extension in dialplan/default.xml to receive bridge incoming calls to my skype instance. Frankly speaking, I'm not sure whether this definition is correct. How should I define the expression? When I dial the telephone number, the FS console has no response and I hear nother but busy tones. For outgoing calls, I tried something like this: originate openzap/1/1/xxxxxxxx &echo, while "xxxxxxxx" is my handphone number. Again, my handphone has no response. Hopefully I've explained my situation clearly. Please kindly enlighten where the problem might be. Thanks, -Jingwei p.s. here is the outgoing log trace for your reference. freeswitch at localhost.localdomain> originate openzap/1/1/xxxxxxxx &echo 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound channel OpenZAP/1:1/xxxxxxxx 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel OpenZAP/1:1/xxxxxxxx [6f843194-18ce-4525-862f-f5f4e96db5eb] 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 (OpenZAP/1:1/xxxxxxxx) State Change CS_NEW -> CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 from DOWN to DIALING 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/xxxxxxxx) State Change CS_INIT -> CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/xxxxxxxx CHANNEL ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/1:1/xxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA going to sleep 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_HOME ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_OFFICE ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_MOBILE ||| 2009-12-03 17:21:34.684709 [DEBUG] ozmod_analog.c:340 Changing state on 1:1 from DIALING to BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:579 Changing state on 1:1 from BUSY to DOWN 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DOWN 2009-12-03 17:21:34.724706 [DEBUG] mod_openzap.c:1334 got FXO sig 1:1 [STOP] 2009-12-03 17:21:34.724706 [NOTICE] mod_openzap.c:1352 Hangup OpenZAP/1:1/xxxxxxxx [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.724706 [DEBUG] switch_channel.c:1912 Send signal OpenZAP/1:1/xxxxxxxx [KILL] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_HANGUP 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-12-03 17:21:34.724706 [DEBUG] zap_io.c:1234 channel done 1:1 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:766 ANALOG CHANNEL 1:1 thread ended. 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP API CALL [originate(openzap/1/1/xxxxxxxx &echo)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.724706 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.725762 [WARNING] mod_openzap.c:474 VETO Changing state on 1:1 from DOWN to HANGUP 2009-12-03 17:21:34.725762 [DEBUG] mod_openzap.c:510 OpenZAP/1:1/xxxxxxxx CHANNEL HANGUP 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:1/xxxxxxxx Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:1/xxxxxxxx) State Change CS_HANGUP -> CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:1/xxxxxxxx Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:1/xxxxxxxx) State Change CS_REPORTING -> CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:1136 Session 1 (OpenZAP/1:1/xxxxxxxx) Locked, Waiting on external entities 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1154 Session 1 (OpenZAP/1:1/xxxxxxxx) Ended 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1156 Close Channel OpenZAP/1:1/xxxxxxxx [CS_DESTROY] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:423 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY freeswitch at localhost.localdomain> 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:60 OpenZAP/1:1/xxxxxxxx Standard DESTROY 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/a22e8f5a/attachment-0001.html From b_ball_henry at hotmail.com Thu Dec 3 01:36:46 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 3 Dec 2009 17:36:46 +0800 Subject: [Freeswitch-users] Gateway issue with no audio Message-ID: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. Finally, I gave up play around with the internal profile and put the gateway *settings under external profile. And magically, it worked.* I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? by the way, I am using freeswitch 1.4 stable version. -- Henry Huang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8d14e1f8/attachment.html From devel at thom.fr.eu.org Thu Dec 3 02:16:25 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 11:16:25 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Message-ID: <82fda3419daad21bded177dc2b113396@thom.fr.eu.org> I'm already using the latest wanpipe drivers (3.5.8), so yes. Fran?ois On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitch.org [9] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [10] ClueCon http://www.cluecon.com/ [11] Twitter: http://twitter.com/FreeSWITCH_wire [12] AIM: anthm MSN:anthony_minessale at hotmail.com [13] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [14] IRC: irc.freenode.net [15] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [16] iax:guest at conference.freeswitch.org/888 [17] googletalk:conf+888 at conference.freeswitch.org [18] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [23] ClueCon http://www.cluecon.com/ [24] Twitter: http://twitter.com/FreeSWITCH_wire [25] AIM: anthm MSN:anthony_minessale at hotmail.com [26] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [27] IRC: irc.freenode.net [28] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [29] iax:guest at conference.freeswitch.org/888 [30] googletalk:conf+888 at conference.freeswitch.org [31] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [32] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [33] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [34] http://www.freeswitch.org [35] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [36] ClueCon http://www.cluecon.com/ [37] Twitter: http://twitter.com/FreeSWITCH_wire [38] AIM: anthm MSN:anthony_minessale at hotmail.com [39] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [40] IRC: irc.freenode.net [41] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [42] iax:guest at conference.freeswitch.org/888 [43] googletalk:conf+888 at conference.freeswitch.org [44] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:anthony.minessale at gmail.com [3] mailto:devel at thom.fr.eu.org [4] http://files.freeswitch.org [5] mailto:devel at thom.fr.eu.org [6] mailto:FreeSWITCH-users at lists.freeswitch.org [7] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] http://lists.freeswitch.org/mailman/options/freeswitch-users [9] http://www.freeswitch.org [10] http://www.freeswitch.org/ [11] http://www.cluecon.com/ [12] http://twitter.com/FreeSWITCH_wire [13] mailto:MSN%3Aanthony_minessale at hotmail.com [14] mailto:PAYPAL%3Aanthony.minessale at gmail.com [15] http://irc.freenode.net [16] mailto:sip%3A888 at conference.freeswitch.org [17] http://iax:guest at conference.freeswitch.org/888 [18] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [19] mailto:FreeSWITCH-users at lists.freeswitch.org [20] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21] http://lists.freeswitch.org/mailman/options/freeswitch-users [22] http://www.freeswitch.org [23] http://www.freeswitch.org/ [24] http://www.cluecon.com/ [25] http://twitter.com/FreeSWITCH_wire [26] mailto:MSN%3Aanthony_minessale at hotmail.com [27] mailto:PAYPAL%3Aanthony.minessale at gmail.com [28] http://irc.freenode.net [29] mailto:sip%3A888 at conference.freeswitch.org [30] http://iax:guest at conference.freeswitch.org/888 [31] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [32] mailto:FreeSWITCH-users at lists.freeswitch.org [33] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] http://lists.freeswitch.org/mailman/options/freeswitch-users [35] http://www.freeswitch.org [36] http://www.freeswitch.org/ [37] http://www.cluecon.com/ [38] http://twitter.com/FreeSWITCH_wire [39] mailto:MSN%3Aanthony_minessale at hotmail.com [40] mailto:PAYPAL%3Aanthony.minessale at gmail.com [41] http://irc.freenode.net [42] mailto:sip%3A888 at conference.freeswitch.org [43] http://iax:guest at conference.freeswitch.org/888 [44] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/a9643b6b/attachment.html From devel at thom.fr.eu.org Thu Dec 3 02:17:09 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 11:17:09 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: <0d3ea0a7a75877ecc715076b03899b06@thom.fr.eu.org> Thanks. I did not succed to fincing the correct syntx with inline, but the transfer application did work. Fran?ois On Wed, 2 Dec 2009 12:21:54 -0600, Anthony Minessale wrote: bind to the transfer app so that it transfers the call to the vm extension that way the current application is always interrupted and replaced. The special "inline" dialplan lets you transfer calls right to an application use "inline" as the dp name and voicemail: as the extension On Wed, Dec 2, 2009 at 4:57 AM, Fran?ois Legal wrote: Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale at hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [12] iax:guest at conference.freeswitch.org/888 [13] googletalk:conf+888 at conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:FreeSWITCH-users at lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale at hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale at gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888 at conference.freeswitch.org [13] http://iax:guest at conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c138525c/attachment-0001.html From codecomplete at free.fr Thu Dec 3 04:17:22 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 3 Dec 2009 04:17:22 -0800 (PST) Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? Message-ID: <26625105.post@talk.nabble.com> Hello In a thread back in March, I read that support for IAX in FreeSwitch is a bit of kludge and since there's not much demand for it, chances are it won't improve in the foreseeable future. So I'd like some feedback from users who routinely connect to a FreeSwitch server from various venues, ie. wifi hotspots at McD, Ethernet LAN in hotels, etc. (in my case, the FreeSwitch server is located in a private network behind a NAT router with SIP/RTP ports statically mapped.) Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) ports fail being opened dynamically to work properly, or does SIP today really work well over NAT firewalls? Thank you. -- View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From oscav at hotmail.fr Thu Dec 3 04:21:04 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 3 Dec 2009 04:21:04 -0800 (PST) Subject: [Freeswitch-users] How to run a JS script periodically Message-ID: <26625147.post@talk.nabble.com> Hi, Someone knows how to run periodically a JS script ?? The purpose is to write to a db some global informations (Global Variables) about FS like every 5 minutes. Thanks. -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Dec 3 05:28:57 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 3 Dec 2009 21:28:57 +0800 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <26625147.post@talk.nabble.com> References: <26625147.post@talk.nabble.com> Message-ID: <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> Not sure about js, but in lua, you can use luarun to run a long-running script like loop do sth. sleep 5min end and also it can be set to start with freeswitch in lua.conf.xml I guess you can also use jsrun to run js. And, if you run every 5 min, why not use crontab? fs_cli -x "jsrun xx.js" 2009/12/3 Oscav : > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to write > to a db some global informations (Global Variables) about FS like every 5 > minutes. > > Thanks. > > > -- > View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rob4manhere at gmail.com Thu Dec 3 05:31:16 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 3 Dec 2009 07:31:16 -0600 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <26625147.post@talk.nabble.com> References: <26625147.post@talk.nabble.com> Message-ID: What about cron? Create a cron entry like: */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" But if you're just dumping global variables, you could easily retrieve them directly from fs_cli without running an app and process the output however you'd like: /usr/local/freeswitch/bin/fs_cli -x "global_getvar" On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to > write > to a db some global informations (Global Variables) about FS like every 5 > minutes. > > Thanks. > > > -- > View this message in context: > http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/427638e5/attachment.html From mike at jerris.com Thu Dec 3 05:48:55 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:48:55 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26621005.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> Message-ID: <9A0C4C44-B0D5-44F2-9C85-89FA27A89300@jerris.com> First off, maybe this conversation is better suited to the dev list, and second off, the current setup of where we do timers, where we poll, polling frequency and architecture is the result of 4+ years of ongoing testing and optimization. We have tried all different methods throughout. Sometimes what we found to be most efficient is not what we thought at first would be, but testing showed otherwise. We have always optimized the general case as to if there are many calls, and no suggestion would be implemented that hurts this case. That being said, if you could really come up with a way for this to be more efficient in any case, without sacrificing performance int he other cases, you are able to prove this with extensive test results, and you are able to prove that it does not impact for example call quality in any of the hundreds of edge cases that have led us to the point we are now, then we may be interested in taking such a patch. Mike On Dec 2, 2009, at 11:58 PM, eaf wrote: > > As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it > could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides > that. > > Yeah, there is a global timestamp... It's easy to workaround that for RTP > who calls switch_micro_time_now()... But if somebody accesses > runtime.timestamp directly, it's gonna be tough to grep for that. If only > this was C++... > > I'll play around. Never liked polling too much. Never could've guessed that > polling could be so useful for scalability ;) My naive implementation > would've pulled timestamp via system calls and would've done sleeping by > passing exact interval to select() instead of syncing with a pacing thread. > Which would be dead-quiet at idle time, but, of course, would stop scaling > at some point due to excessive number of system calls. > > Thanks. > > > Michael Jerris wrote: >> >> In short. No, you can not for many reasons. The milisecond tic is >> used throughout the code even when there is not any calls up. You can >> grep for switch_cond_next if you would like to see where but it is >> required to keep our global timestamp and for pacing the scheduler >> among other services that run all the time. >> >> Mike >> >> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >>> Can I reduce resolution of that timer thread 10 times? I mean, I >>> glanced >>> through the code, and see that among others (are there others?) RTP >>> and IVR >>> set up their timers that are subsequently managed by this thread. >>> RTP timers >>> should be eliminated by that setting you've suggested. IVR timers >>> are set at >>> 20ms... So, if the thread is set to wake up every 10ms instead of >>> 1ms it >>> should be able to wake up those IVR timers just fine. Right? >>> >>> That's a cool design to have one dedicated thread that maintains >>> accurate >>> timing and then broadcasts via condition variables to hundreds of >>> other >>> threads events that they can register for. I'm sure it's one of the >>> reasons >>> why FS scales so much better than Asterisk. But for poor low-end >>> setups that >>> sit in the closet, eat only 6W of power and hardly ever run more >>> than two >>> calls at the same time, can I hack it somehow to be more UNIX- >>> friendly? I.e. >>> make it stuck in select() or recv() when there is nothing to do, call >>> clock_gettime() right from the thread that wants and when it wants >>> to know >>> current time? >>> >>> Say, what if that thread is made to suspend on a condition variable >>> in case >>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>> thread comes up and adds its timer into the matrix, it could wake up >>> the >>> timer thread and enjoy accurate timing as needed, on demand? And in- >>> between >>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>> sitting on a wait queue in the kernel is way better than go back and >>> forth >>> incrementing counters that nobody even needs at the moment? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> idle is a 4 letter word to a realtime application. >>>> >>>> The core keeps a single high-priority thread to keep 1ms timing and >>>> expands >>>> that broadcasting >>>> to hundreds or thousand of threads who need accurate timing. >>>> >>>> Your choppy audio is caused by linksys lying about the packet len >>>> that >>>> it's >>>> using and we set our timer >>>> to the wrong speed. >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26621005.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Dec 3 05:50:19 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:50:19 -0500 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: References: <26625147.post@talk.nabble.com> Message-ID: <53E931F0-43C6-4756-B636-F17D7E076A1B@jerris.com> You could also use the scheduler to run the jsrun command inside FreeSWITCH. Mike On Dec 3, 2009, at 8:31 AM, Rob Forman wrote: > What about cron? > > Create a cron entry like: > */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" > > But if you're just dumping global variables, you could easily retrieve them directly from fs_cli without running an app and process the output however you'd like: > > /usr/local/freeswitch/bin/fs_cli -x "global_getvar" > > > On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to write > to a db some global informations (Global Variables) about FS like every 5 > minutes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/70f1c40f/attachment.html From mike at jerris.com Thu Dec 3 05:57:02 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:57:02 -0500 Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <26613841.post@talk.nabble.com> References: <26610094.post@talk.nabble.com> <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> <26613841.post@talk.nabble.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#originate Usage: originate |&() [] [] [] [] [] You can do this via shelling out to fs_cli like your example below or using esl directly from php: http://wiki.freeswitch.org/wiki/Esl Mike On Dec 2, 2009, at 1:23 PM, eaf wrote: > > I need a way to start a call from the PHP script to the originating number, > tell the party on that number to hold on, start another call to destination > number, and bridge everything together. On both legs I need to pass custom > caller ID. I can of course open direct connections to VOIP gateways right > from PHP, but I want to reuse existing routing rules in the dial plan, hence > I want to know what's the best way of making originate go through a specific > context of the dial plan. > > As for the number of calls per second, it's going to be only occasionally > used. > > > mercutioviz wrote: >> >> On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: >> >>> >>> What would be the best way of making originate() run call through a dial >>> plan >>> (compared to directly going to a specified VOIP gateway). Would it be >>> loopbacks, i.e. smth like this? >>> >>> /opt/freeswitch/bin/fs_cli -x "originate >>> >>> {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML >>> '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js >>> zzzzzzzzzz)'" >>> >>> The idea of this is that originate() sets up the first call, then >>> webcall.js >>> plays back a WAV, and bridges the first call with the second one (also >>> set >>> up via loopback). >>> >>> >> Could you describe the problem that you're trying to solve? That would >> make >> it easier to know if what you've come up with is the best solution. How >> many >> calls per second were you wanting to generate with this setup? >> -MC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26613841.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Dec 3 06:08:04 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 09:08:04 -0500 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03c401ca73bf$1cea8600$56bf9200$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: The behavior is probably expected, the unhelpful error is probably undesirable but it would make a mess of the dial-plan to clean that up. Mike On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote: > Is this reasonable given it was the only call in FreeSwitch at the time? How > can this situation be corrected in the future? > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Wednesday, December 02, 2009 3:35 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Eavesdrop error? > > it probably just means the uuid was not retrieved from the db when you > called the eavesdrop exten which does the lookup on the uuid for the hash > key based on what ext you hit to retrieve the most recent uuid that called > that ext. > > > On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > Sorry, svn 15753 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb > Sent: Wednesday, December 02, 2009 2:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Eavesdrop error? > > I tried to use eavesdrop today and it did not work. The error message in the > log is: > > [ERR] mod_dptools.c:334 Usage: [all | ] > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > incorrect? > > http://pastebin.freeswitch.org/11363 > > Thanks Lars > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From erandr-junk at usa.net Thu Dec 3 06:17:54 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 06:17:54 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26621005.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> Message-ID: <26626634.post@talk.nabble.com> Oh, it's not just one timer thread... Why, why is sql_thread keeps on checking for messages every millisecond? Couldn't there be some signalling implemented that will make the thread suspend on condition variable or a socket/pipe in between? #0 do_sleep (t=1000) at src/switch_time.c:109 #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at src/switch_core_sqldb.c:783 Why does this sofia_profile_worker_thread keeps on looping checking for the queue? Have a semaphore! #0 do_sleep (t=1000) at src/switch_time.c:109 #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, obj=0x80f2490) at sofia.c:978 Nothing's happening on the box, but there are three threads that pretend to be actively busy with smth. Others at least sleep for hundreds of milliseconds, not for one. And there is even infrastructure present to do blocking pops: i.e. why couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed with 1ms sleeps? This looping is such a waste... eaf wrote: > > As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, > it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > overrides that. > > Yeah, there is a global timestamp... It's easy to workaround that for RTP > who calls switch_micro_time_now()... But if somebody accesses > runtime.timestamp directly, it's gonna be tough to grep for that. If only > this was C++... > > I'll play around. Never liked polling too much. Never could've guessed > that polling could be so useful for scalability ;) My naive implementation > would've pulled timestamp via system calls and would've done sleeping by > passing exact interval to select() instead of syncing with a pacing > thread. Which would be dead-quiet at idle time, but, of course, would stop > scaling at some point due to excessive number of system calls. > > Thanks. > > > Michael Jerris wrote: >> >> In short. No, you can not for many reasons. The milisecond tic is >> used throughout the code even when there is not any calls up. You can >> grep for switch_cond_next if you would like to see where but it is >> required to keep our global timestamp and for pacing the scheduler >> among other services that run all the time. >> >> Mike >> >> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >>> Can I reduce resolution of that timer thread 10 times? I mean, I >>> glanced >>> through the code, and see that among others (are there others?) RTP >>> and IVR >>> set up their timers that are subsequently managed by this thread. >>> RTP timers >>> should be eliminated by that setting you've suggested. IVR timers >>> are set at >>> 20ms... So, if the thread is set to wake up every 10ms instead of >>> 1ms it >>> should be able to wake up those IVR timers just fine. Right? >>> >>> That's a cool design to have one dedicated thread that maintains >>> accurate >>> timing and then broadcasts via condition variables to hundreds of >>> other >>> threads events that they can register for. I'm sure it's one of the >>> reasons >>> why FS scales so much better than Asterisk. But for poor low-end >>> setups that >>> sit in the closet, eat only 6W of power and hardly ever run more >>> than two >>> calls at the same time, can I hack it somehow to be more UNIX- >>> friendly? I.e. >>> make it stuck in select() or recv() when there is nothing to do, call >>> clock_gettime() right from the thread that wants and when it wants >>> to know >>> current time? >>> >>> Say, what if that thread is made to suspend on a condition variable >>> in case >>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>> thread comes up and adds its timer into the matrix, it could wake up >>> the >>> timer thread and enjoy accurate timing as needed, on demand? And in- >>> between >>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>> sitting on a wait queue in the kernel is way better than go back and >>> forth >>> incrementing counters that nobody even needs at the moment? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> idle is a 4 letter word to a realtime application. >>>> >>>> The core keeps a single high-priority thread to keep 1ms timing and >>>> expands >>>> that broadcasting >>>> to hundreds or thousand of threads who need accurate timing. >>>> >>>> Your choppy audio is caused by linksys lying about the packet len >>>> that >>>> it's >>>> using and we set our timer >>>> to the wrong speed. >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26626634.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From abeka at greatiam.com Thu Dec 3 06:22:38 2009 From: abeka at greatiam.com (Otis) Date: Thu, 03 Dec 2009 14:22:38 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> Message-ID: <4B17C9AE.2010408@greatiam.com> Michael Collins wrote: > > > On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle > wrote: > > On Wed, Dec 02, Otis wrote: > Snip... > > > Thanks. > > > > I would like all extensions on say server A to be contactable > by those > > on server B and vice versa. > > The example I gave you should get you started. Let us know how > you get along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > > Remember, too, that gateways are useful for doing auth/reg so having a > gateway on each box that registers to the other box is pretty handy. > If you run into any trouble trying to set it up you can ask here or > join us in #freeswitch on irc.freenode.net . > -MC Hi FC I used your code : replacing with my box's ip address. I have received any errors in the fs_cli console neither is there any reference to my box'x ipddress. Any way to check all is well ? And how do I join join us in #freeswitch on irc.freenode.net . ? Went to the freenode.net site and got lost. Will persevere. Thanks From william.suffill at gmail.com Thu Dec 3 06:34:44 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 3 Dec 2009 09:34:44 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B17C9AE.2010408@greatiam.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> <4B17C9AE.2010408@greatiam.com> Message-ID: <6b65470d0912030634s4c9a304fu52ded127f98760a@mail.gmail.com> http://www.freeswitch.org/ On the right side. Join IRC Just fill in a nickname and click JOIN IRC -- W From erandr-junk at usa.net Thu Dec 3 06:55:21 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 06:55:21 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26626634.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> Message-ID: <26627246.post@talk.nabble.com> Btw, I have these popping up in my logs from time to time: 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration Detected! Syncing Clock In this case an incoming call rang to both FS and Asterisk, Asterisk picked up, but the surge of activity made FS timer thread miss a beat or two. eaf wrote: > > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > checking for messages every millisecond? Couldn't there be some signalling > implemented that will make the thread suspend on condition variable or a > socket/pipe in between? > > #0 do_sleep (t=1000) at src/switch_time.c:109 > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at > src/switch_core_sqldb.c:783 > > Why does this sofia_profile_worker_thread keeps on looping checking for > the queue? Have a semaphore! > > #0 do_sleep (t=1000) at src/switch_time.c:109 > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > obj=0x80f2490) at sofia.c:978 > > Nothing's happening on the box, but there are three threads that pretend > to be actively busy with smth. Others at least sleep for hundreds of > milliseconds, not for one. > > And there is even infrastructure present to do blocking pops: i.e. why > couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed > with 1ms sleeps? This looping is such a waste... > > > eaf wrote: >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> overrides that. >> >> Yeah, there is a global timestamp... It's easy to workaround that for RTP >> who calls switch_micro_time_now()... But if somebody accesses >> runtime.timestamp directly, it's gonna be tough to grep for that. If only >> this was C++... >> >> I'll play around. Never liked polling too much. Never could've guessed >> that polling could be so useful for scalability ;) My naive >> implementation would've pulled timestamp via system calls and would've >> done sleeping by passing exact interval to select() instead of syncing >> with a pacing thread. Which would be dead-quiet at idle time, but, of >> course, would stop scaling at some point due to excessive number of >> system calls. >> >> Thanks. >> >> >> Michael Jerris wrote: >>> >>> In short. No, you can not for many reasons. The milisecond tic is >>> used throughout the code even when there is not any calls up. You can >>> grep for switch_cond_next if you would like to see where but it is >>> required to keep our global timestamp and for pacing the scheduler >>> among other services that run all the time. >>> >>> Mike >>> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >>> >>>> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >>>> glanced >>>> through the code, and see that among others (are there others?) RTP >>>> and IVR >>>> set up their timers that are subsequently managed by this thread. >>>> RTP timers >>>> should be eliminated by that setting you've suggested. IVR timers >>>> are set at >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >>>> 1ms it >>>> should be able to wake up those IVR timers just fine. Right? >>>> >>>> That's a cool design to have one dedicated thread that maintains >>>> accurate >>>> timing and then broadcasts via condition variables to hundreds of >>>> other >>>> threads events that they can register for. I'm sure it's one of the >>>> reasons >>>> why FS scales so much better than Asterisk. But for poor low-end >>>> setups that >>>> sit in the closet, eat only 6W of power and hardly ever run more >>>> than two >>>> calls at the same time, can I hack it somehow to be more UNIX- >>>> friendly? I.e. >>>> make it stuck in select() or recv() when there is nothing to do, call >>>> clock_gettime() right from the thread that wants and when it wants >>>> to know >>>> current time? >>>> >>>> Say, what if that thread is made to suspend on a condition variable >>>> in case >>>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>>> thread comes up and adds its timer into the matrix, it could wake up >>>> the >>>> timer thread and enjoy accurate timing as needed, on demand? And in- >>>> between >>>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>>> sitting on a wait queue in the kernel is way better than go back and >>>> forth >>>> incrementing counters that nobody even needs at the moment? >>>> >>>> >>>> Anthony Minessale-2 wrote: >>>>> >>>>> idle is a 4 letter word to a realtime application. >>>>> >>>>> The core keeps a single high-priority thread to keep 1ms timing and >>>>> expands >>>>> that broadcasting >>>>> to hundreds or thousand of threads who need accurate timing. >>>>> >>>>> Your choppy audio is caused by linksys lying about the packet len >>>>> that >>>>> it's >>>>> using and we set our timer >>>>> to the wrong speed. >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rupa at rupa.com Thu Dec 3 07:00:01 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 3 Dec 2009 09:00:01 -0600 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> References: <26625147.post@talk.nabble.com> <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> Message-ID: If doing this, I'd suggest checking for a global var to see if the script should terminate itself. Otherwise, you'll have to bring down the whole freeswitch to stop this script. On Thu, Dec 3, 2009 at 7:28 AM, Seven Du wrote: > Not sure about js, but in lua, you can use luarun to run a > long-running script like > > > loop > do sth. > sleep 5min > end > > and also it can be set to start with freeswitch in lua.conf.xml > > I guess you can also use jsrun to run js. > > And, if you run every 5 min, why not use crontab? > > fs_cli -x "jsrun xx.js" > > > 2009/12/3 Oscav : > > > > Hi, > > > > Someone knows how to run periodically a JS script ?? The purpose is to > write > > to a db some global informations (Global Variables) about FS like every 5 > > minutes. > > > > Thanks. > > > > > > -- > > View this message in context: > http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/950f5139/attachment.html From testeador01 at gmail.com Thu Dec 3 07:30:48 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 3 Dec 2009 10:30:48 -0500 Subject: [Freeswitch-users] Call transfer got broken for me Message-ID: Hello, It was all ok until yesterday when i updated to svn 15761(last update before that was about 4 days ago), Now I have this issue: someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext 200 200 picks up, then 200 transfers the call to 205 call gets lost (it used to transfer normal until the moment I updated) Today I updated to 15771 and the issue is still there. Can anyone help me figure out what is going on? Call log: http://pastebin.freeswitch.org/11374 thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/15d36757/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 3 07:56:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 09:56:14 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> Message-ID: <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> Try trunk again On Wed, Dec 2, 2009 at 5:33 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I am not sure what you are sending over the socket but you have a queued > hangup being processed on line 640 of your pastebin > are you executing any commands with a ! character in it by any chance or > executing the hangup app on purpose? > > > > > On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> Tony, >> >> Thanks for that but now it appears that the call just gets hung up >> on when the caller takes the callee off hold. Debug here: >> >> http://pastebin.freeswitch.org/11359 >> >> Thanks again! >> >> On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale >> wrote: >> > I decided to just change the code so its more elegant to handle >> recursive >> > broadcasting so you can try again and see if that helps. >> > >> > >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/fbf05b86/attachment.html From mike at jerris.com Thu Dec 3 07:59:44 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 10:59:44 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? Mike On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote: > I have read some of the archived emails about HA, loadbalancing, > failover etc and I am still a bit confused about how I could set up > some sort of resiliency with freeswitch. > > My situation is much less complex than the scenarios people were > talking about and I hoping the solution is similarly much less > complex. > > I have two machines. Both will run freeswitch and also an IVR > application with local databases. I will take care of the database, > application and configuration synchronization between the two > machines. Ideally the calls would be load balanced between the > machines and if any application falls down then the calls should go to > the other machine. Same if I take a machine down for whatever reason. > > If a machine goes down I am willing to "lose" those people who were > making a call at the time. I do have a flag in the application which > will stop answering the calls while processing the existing calls for > a graceful shutdown and hopefully the load balancer would shuttle the > calls to the other machine while this is happening. > > At this stage everything is done via SIP. > > My questions are... > > Do I have to have a sip proxy? If the answer is yes it seems like I > have to set up two sip proxies so I don't have another single point of > failure. Can I load the sip proxies on the same machine? Do I need two > more machines? > > If I take load balancing out of the picture would it be possible to do > a simple linux HA or a windows built in ip failover solution? Would a > simple IP failover work over UDP or would I have to use IAX and tcp/ip > ? > > Is it better to go the virtualization route? > > Sorry if these are dumb questions. I am just trying to get my head > wrapped around this. I don't need five nines (although that would be > awesome), I just want a reasonable degree of assurance that my app can > keep taking calls in case something weird happens. From mike at jerris.com Thu Dec 3 08:01:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:01:34 -0500 Subject: [Freeswitch-users] can't register Inphonex In-Reply-To: <005a01ca73c9$bc2dcf60$34896e20$@ca> References: <005a01ca73c9$bc2dcf60$34896e20$@ca> Message-ID: <30C82C4E-00FD-45BE-9D45-93FD0F99694E@jerris.com> You can turn up the full freeswitch debug or enable the siptrace on the sip profile to get more information about this. This looks like a nat related issue getting no response from the provider. A sip trace is probably the best tool to figure this one out. sofia profile internal siptrace on Mike On Dec 2, 2009, at 10:35 PM, John Lalande wrote: > I am new to FS having ditched Asterisk a few weeks ago. I have iptel registered but I cannot get Inphonex to work. I am using the settings fromhttp://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no avail. > > The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout [408]." > > Is there some way to debug this? sofia status displays: > > Name Type Data State > ================================================================================================= > external profile sip:mod_sofia at 192.168.125.15:5080 RUNNING (0) > example.com gateway sip:joeuser at example.com NOREG > inphonex gateway sip:5285418 at sip.inphonex.com FAILED (retry: 28s) > iptel gateway sip:jlalande at sip.iptel.org REGED > internal profile sip:mod_sofia at 192.168.125.15:5060 RUNNING (0) > internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) > 192.168.125.15 alias internal ALIASED > ================================================================================================= > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/937cd8cb/attachment.html From mike at jerris.com Thu Dec 3 08:07:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:07:34 -0500 Subject: [Freeswitch-users] Gateway issue with no audio In-Reply-To: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> References: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> Message-ID: You may want to try this again with latest svn trunk. We have done quite a lot of work to make nat support much better sense 1.0.4 Mike p.s. I can't comment about version 1.4 due to broken flux capacitor. On Dec 3, 2009, at 4:36 AM, Henry Huang wrote: > My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. > > Finally, I gave up play around with the internal profile and put the gateway settings under external profile. And magically, it worked. I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? > > by the way, I am using freeswitch 1.4 stable version. > > > > -- > Henry Huang > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/aaabc8cd/attachment-0001.html From mike at jerris.com Thu Dec 3 08:10:59 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:10:59 -0500 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <26625105.post@talk.nabble.com> References: <26625105.post@talk.nabble.com> Message-ID: <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> with the right clients, it nearly always works well. with a client that does not support stun or at least rfc 3581 the results are much more sketchy and require more hacks on the server side, but with enough effort can almost always be made to work. Mike On Dec 3, 2009, at 7:17 AM, Fred-145 wrote: > > Hello > > In a thread back in March, I read that support for IAX in FreeSwitch is a > bit of kludge and since there's not much demand for it, chances are it won't > improve in the foreseeable future. > > So I'd like some feedback from users who routinely connect to a FreeSwitch > server from various venues, ie. wifi hotspots at McD, Ethernet LAN in > hotels, etc. (in my case, the FreeSwitch server is located in a private > network behind a NAT router with SIP/RTP ports statically mapped.) > > Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) > ports fail being opened dynamically to work properly, or does SIP today > really work well over NAT firewalls? > > Thank you. > -- > View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 3 08:16:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:16:26 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26627246.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> Message-ID: <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> If you see that message then your machine/os/combo is having some problems keeping up. It's not the timer missing anything its the monotonic clock detecting a 1 second or more differential from what its next prediction for the time should be. The best way to trigger this would be to suspend FS with control-z or attach to it with gdb blocking the entire process, that 1ms thread would have to miss 1000 iterations to trigger that warning. Btw, that error message is at line 471 not 473 so you are using modified code. Its possible your box has a bad monotonic timer, you can set under in switch.conf.xml We are now starting to guess you are using some small embedded type platform perhaps? I've run FS even on a nokia n810 and never caused that message to fire. if 1 call can interrupt the cpu enough to cause noticeable issues you might want to consider running the process at a greater priority by using the -hp command line arg or at least nice it Why don't you tell us the whole story about what OS/platform you are using here rather that form conjectures about what is wrong with our code that thousands of people are happy with. On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > > Btw, I have these popping up in my logs from time to time: > > 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > Detected! Syncing Clock > > In this case an incoming call rang to both FS and Asterisk, Asterisk picked > up, but the surge of activity made FS timer thread miss a beat or two. > > > eaf wrote: > > > > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > > checking for messages every millisecond? Couldn't there be some > signalling > > implemented that will make the thread suspend on condition variable or a > > socket/pipe in between? > > > > #0 do_sleep (t=1000) at src/switch_time.c:109 > > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at > > src/switch_core_sqldb.c:783 > > > > Why does this sofia_profile_worker_thread keeps on looping checking for > > the queue? Have a semaphore! > > > > #0 do_sleep (t=1000) at src/switch_time.c:109 > > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > > obj=0x80f2490) at sofia.c:978 > > > > Nothing's happening on the box, but there are three threads that pretend > > to be actively busy with smth. Others at least sleep for hundreds of > > milliseconds, not for one. > > > > And there is even infrastructure present to do blocking pops: i.e. why > > couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed > > with 1ms sleeps? This looping is such a waste... > > > > > > eaf wrote: > >> > >> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, > >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> overrides that. > >> > >> Yeah, there is a global timestamp... It's easy to workaround that for > RTP > >> who calls switch_micro_time_now()... But if somebody accesses > >> runtime.timestamp directly, it's gonna be tough to grep for that. If > only > >> this was C++... > >> > >> I'll play around. Never liked polling too much. Never could've guessed > >> that polling could be so useful for scalability ;) My naive > >> implementation would've pulled timestamp via system calls and would've > >> done sleeping by passing exact interval to select() instead of syncing > >> with a pacing thread. Which would be dead-quiet at idle time, but, of > >> course, would stop scaling at some point due to excessive number of > >> system calls. > >> > >> Thanks. > >> > >> > >> Michael Jerris wrote: > >>> > >>> In short. No, you can not for many reasons. The milisecond tic is > >>> used throughout the code even when there is not any calls up. You can > >>> grep for switch_cond_next if you would like to see where but it is > >>> required to keep our global timestamp and for pacing the scheduler > >>> among other services that run all the time. > >>> > >>> Mike > >>> > >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >>> > >>>> > >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >>>> glanced > >>>> through the code, and see that among others (are there others?) RTP > >>>> and IVR > >>>> set up their timers that are subsequently managed by this thread. > >>>> RTP timers > >>>> should be eliminated by that setting you've suggested. IVR timers > >>>> are set at > >>>> 20ms... So, if the thread is set to wake up every 10ms instead of > >>>> 1ms it > >>>> should be able to wake up those IVR timers just fine. Right? > >>>> > >>>> That's a cool design to have one dedicated thread that maintains > >>>> accurate > >>>> timing and then broadcasts via condition variables to hundreds of > >>>> other > >>>> threads events that they can register for. I'm sure it's one of the > >>>> reasons > >>>> why FS scales so much better than Asterisk. But for poor low-end > >>>> setups that > >>>> sit in the closet, eat only 6W of power and hardly ever run more > >>>> than two > >>>> calls at the same time, can I hack it somehow to be more UNIX- > >>>> friendly? I.e. > >>>> make it stuck in select() or recv() when there is nothing to do, call > >>>> clock_gettime() right from the thread that wants and when it wants > >>>> to know > >>>> current time? > >>>> > >>>> Say, what if that thread is made to suspend on a condition variable > >>>> in case > >>>> if there are no timers registered in TIMER_MATRIX? Then, if some other > >>>> thread comes up and adds its timer into the matrix, it could wake up > >>>> the > >>>> timer thread and enjoy accurate timing as needed, on demand? And in- > >>>> between > >>>> the calls, when there is no RTP or IVR, it will all go silent? I mean, > >>>> sitting on a wait queue in the kernel is way better than go back and > >>>> forth > >>>> incrementing counters that nobody even needs at the moment? > >>>> > >>>> > >>>> Anthony Minessale-2 wrote: > >>>>> > >>>>> idle is a 4 letter word to a realtime application. > >>>>> > >>>>> The core keeps a single high-priority thread to keep 1ms timing and > >>>>> expands > >>>>> that broadcasting > >>>>> to hundreds or thousand of threads who need accurate timing. > >>>>> > >>>>> Your choppy audio is caused by linksys lying about the packet len > >>>>> that > >>>>> it's > >>>>> using and we set our timer > >>>>> to the wrong speed. > >>>>> > >>>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/061f041e/attachment.html From mike at jerris.com Thu Dec 3 08:21:43 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:21:43 -0500 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: References: Message-ID: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> what revision were you at prior to upgrade or can you narrow the range of versions that broke this any more (or even better the exact version that broke this). Please post this bug to http://jira.freeswitch.org. Mike On Dec 3, 2009, at 10:30 AM, Milena wrote: > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/5e7c3a92/attachment-0001.html From testeador01 at gmail.com Thu Dec 3 08:29:36 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 3 Dec 2009 11:29:36 -0500 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> References: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> Message-ID: This got fixed in version 15773, thank you very much 2009/12/3 Michael Jerris > what revision were you at prior to upgrade or can you narrow the range of > versions that broke this any more (or even better the exact version that > broke this). Please post this bug to http://jira.freeswitch.org. > > Mike > > On Dec 3, 2009, at 10:30 AM, Milena wrote: > > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update > before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext > 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/f5254fa5/attachment.html From anthony.minessale at gmail.com Thu Dec 3 08:32:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:32:19 -0600 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> References: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> Message-ID: <191c3a030912030832o4af484afrad086b848d0c6384@mail.gmail.com> to late it's fixed now. On Thu, Dec 3, 2009 at 10:21 AM, Michael Jerris wrote: > what revision were you at prior to upgrade or can you narrow the range of > versions that broke this any more (or even better the exact version that > broke this). Please post this bug to http://jira.freeswitch.org. > > Mike > > On Dec 3, 2009, at 10:30 AM, Milena wrote: > > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update > before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext > 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/368a0e12/attachment.html From anthony.minessale at gmail.com Thu Dec 3 08:35:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:35:56 -0600 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: <191c3a030912030835s3884a2fal73cf9527041e023b@mail.gmail.com> you could check if the uuid is blank with an expression and playback an audio warning that it's an invalid call. On Thu, Dec 3, 2009 at 8:08 AM, Michael Jerris wrote: > The behavior is probably expected, the unhelpful error is probably > undesirable but it would make a mess of the dial-plan to clean that up. > > Mike > > On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote: > > > Is this reasonable given it was the only call in FreeSwitch at the time? > How > > can this situation be corrected in the future? > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: Wednesday, December 02, 2009 3:35 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Eavesdrop error? > > > > it probably just means the uuid was not retrieved from the db when you > > called the eavesdrop exten which does the lookup on the uuid for the hash > > key based on what ext you hit to retrieve the most recent uuid that > called > > that ext. > > > > > > On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > > Sorry, svn 15753 > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars > Zeb > > Sent: Wednesday, December 02, 2009 2:08 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Eavesdrop error? > > > > I tried to use eavesdrop today and it did not work. The error message in > the > > log is: > > > > [ERR] mod_dptools.c:334 Usage: [all | ] > > > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > > incorrect? > > > > http://pastebin.freeswitch.org/11363 > > > > Thanks Lars > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/44620e04/attachment.html From lists at redbonez.net Thu Dec 3 08:56:23 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 3 Dec 2009 09:56:23 -0700 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <012801ca7439$8cc10ba0$a64322e0$@net> Have you checked out Redfone? While I haven't attempted to implement it yet, my Redfone foneBridge2 claims to be able to handle load balancing and failover between two Asterisk/Freeswitch servers. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Uckun Sent: Wednesday, December 02, 2009 5:50 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] HA questions. I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to "lose" those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From erandr-junk at usa.net Thu Dec 3 09:29:46 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 09:29:46 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> Message-ID: <26629856.post@talk.nabble.com> I'm sorry if I sounded that way. Did mean to. :) Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm Line offset difference is due to some minor logging changes I made to see who's allocating timers and how often. This way I found MOH streaming and that RTP still allocates timers even when it's set to none in the profile. I feel that this platform turned out to be underpowered for FS because it cannot meet its scheduling expectations. I guess, some degree of kernel tweaking or setting priorities will fix that. Meanwhile I just got rid of the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms thread in two (one blocked and waiting for new commands in the SQL queue, the other one checking registrations and gateways with 1sec interval), and don't know yet what to do about the timer thread. Again, I apologize for stupid or accusing questions, I'm just trying to see how FS can be made friendlier to this board. Or the board be made friendlier to FS ;) Anthony Minessale-2 wrote: > > If you see that message then your machine/os/combo is having some problems > keeping up. > It's not the timer missing anything its the monotonic clock detecting a 1 > second or more differential from what its next prediction for the time > should be. The best way to trigger this would be to suspend FS with > control-z or attach to it with gdb blocking the entire process, that 1ms > thread would have to miss 1000 iterations to trigger that warning. > > Btw, that error message is at line 471 not 473 so you are using modified > code. > > Its possible your box has a bad monotonic timer, you can set > > > > under in switch.conf.xml > > We are now starting to guess you are using some small embedded type > platform > perhaps? > I've run FS even on a nokia n810 and never caused that message to fire. > > if 1 call can interrupt the cpu enough to cause noticeable issues you > might > want to consider running the process at a > greater priority by using the -hp command line arg or at least nice it > > Why don't you tell us the whole story about what OS/platform you are using > here rather that form conjectures about what is wrong with our code that > thousands of people are happy with. > > > > > > > > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > >> >> Btw, I have these popping up in my logs from time to time: >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >> Detected! Syncing Clock >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk >> picked >> up, but the surge of activity made FS timer thread miss a beat or two. >> >> >> eaf wrote: >> > >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps on >> > checking for messages every millisecond? Couldn't there be some >> signalling >> > implemented that will make the thread suspend on condition variable or >> a >> > socket/pipe in between? >> > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) >> at >> > src/switch_core_sqldb.c:783 >> > >> > Why does this sofia_profile_worker_thread keeps on looping checking for >> > the queue? Have a semaphore! >> > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >> > obj=0x80f2490) at sofia.c:978 >> > >> > Nothing's happening on the box, but there are three threads that >> pretend >> > to be actively busy with smth. Others at least sleep for hundreds of >> > milliseconds, not for one. >> > >> > And there is even infrastructure present to do blocking pops: i.e. why >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() >> intermixed >> > with 1ms sleeps? This looping is such a waste... >> > >> > >> > eaf wrote: >> >> >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). >> Yes, >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> >> overrides that. >> >> >> >> Yeah, there is a global timestamp... It's easy to workaround that for >> RTP >> >> who calls switch_micro_time_now()... But if somebody accesses >> >> runtime.timestamp directly, it's gonna be tough to grep for that. If >> only >> >> this was C++... >> >> >> >> I'll play around. Never liked polling too much. Never could've guessed >> >> that polling could be so useful for scalability ;) My naive >> >> implementation would've pulled timestamp via system calls and would've >> >> done sleeping by passing exact interval to select() instead of syncing >> >> with a pacing thread. Which would be dead-quiet at idle time, but, of >> >> course, would stop scaling at some point due to excessive number of >> >> system calls. >> >> >> >> Thanks. >> >> >> >> >> >> Michael Jerris wrote: >> >>> >> >>> In short. No, you can not for many reasons. The milisecond tic is >> >>> used throughout the code even when there is not any calls up. You >> can >> >>> grep for switch_cond_next if you would like to see where but it is >> >>> required to keep our global timestamp and for pacing the scheduler >> >>> among other services that run all the time. >> >>> >> >>> Mike >> >>> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >> >>>> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >> >>>> glanced >> >>>> through the code, and see that among others (are there others?) RTP >> >>>> and IVR >> >>>> set up their timers that are subsequently managed by this thread. >> >>>> RTP timers >> >>>> should be eliminated by that setting you've suggested. IVR timers >> >>>> are set at >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >> >>>> 1ms it >> >>>> should be able to wake up those IVR timers just fine. Right? >> >>>> >> >>>> That's a cool design to have one dedicated thread that maintains >> >>>> accurate >> >>>> timing and then broadcasts via condition variables to hundreds of >> >>>> other >> >>>> threads events that they can register for. I'm sure it's one of the >> >>>> reasons >> >>>> why FS scales so much better than Asterisk. But for poor low-end >> >>>> setups that >> >>>> sit in the closet, eat only 6W of power and hardly ever run more >> >>>> than two >> >>>> calls at the same time, can I hack it somehow to be more UNIX- >> >>>> friendly? I.e. >> >>>> make it stuck in select() or recv() when there is nothing to do, >> call >> >>>> clock_gettime() right from the thread that wants and when it wants >> >>>> to know >> >>>> current time? >> >>>> >> >>>> Say, what if that thread is made to suspend on a condition variable >> >>>> in case >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some >> other >> >>>> thread comes up and adds its timer into the matrix, it could wake up >> >>>> the >> >>>> timer thread and enjoy accurate timing as needed, on demand? And in- >> >>>> between >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I >> mean, >> >>>> sitting on a wait queue in the kernel is way better than go back and >> >>>> forth >> >>>> incrementing counters that nobody even needs at the moment? >> >>>> >> >>>> >> >>>> Anthony Minessale-2 wrote: >> >>>>> >> >>>>> idle is a 4 letter word to a realtime application. >> >>>>> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing and >> >>>>> expands >> >>>>> that broadcasting >> >>>>> to hundreds or thousand of threads who need accurate timing. >> >>>>> >> >>>>> Your choppy audio is caused by linksys lying about the packet len >> >>>>> that >> >>>>> it's >> >>>>> using and we set our timer >> >>>>> to the wrong speed. >> >>>>> >> >>>>> >> >>>> >> >>>> -- >> >>>> View this message in context: >> >>>> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>>> users >> >>>> http://www.freeswitch.org >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >>> >> >> >> >> >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristian.kielhofner at gmail.com Thu Dec 3 09:33:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 3 Dec 2009 12:33:50 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> Message-ID: <2d9149cd0912030933k110a89e2j12a8d44bfcb86bbb@mail.gmail.com> Tony, The call no longer hangs up but we still only get hold music in one direction - if the callee places the caller on hold there is no music. PB here: http://pastebin.freeswitch.org/11378 This was on rev 15773. Thanks again Tony! On Thu, Dec 3, 2009 at 10:56 AM, Anthony Minessale wrote: > Try trunk again > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Thu Dec 3 09:44:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 3 Dec 2009 12:44:27 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: <2d9149cd0912030944g5a0b1145q1c292e9bd1e4a11d@mail.gmail.com> I don't think it's the board itself... We have extensively tested FreeSwitch (no modifications) on that exact board with AstLinux and have it running at multiple customer locations. No timing errors, no warnings or errors of any kind. Pretty standard really just don't expect too much from the LX800 (transcoding, resampling, massive numbers of calls, etc). On Thu, Dec 3, 2009 at 12:29 PM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made friendlier > to FS ;) > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From abeka at greatiam.com Thu Dec 3 09:46:07 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 17:46:07 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing Message-ID: <4B17F95F.2000108@greatiam.com> I have copied 1001.xml in directory/default to a test user 2319.xm changing or instances of 1001 in the file to 2319. I then went into default.xml in directory folder and in one of the groups just mimicked 1001 details by changing 1001 to 2319. Connecting to FS gives Forbidden message. However 1001 connects without a problem. What have I missed ? Is there a place that just puts things in do this and that and that to create a new user ? Thanks From mike at jerris.com Thu Dec 3 09:50:26 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 12:50:26 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: I know people with hardware out there in production based on arm11 and those are pretty small processors, not sure how they compare to this. In regards to the DISABLE_1MS_COND, try getting rid of that, it did increase performance on the high end but may be better for you on the low end with lower compute on idle busy loops. Mike On Dec 3, 2009, at 12:29 PM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made friendlier > to FS ;) > > > Anthony Minessale-2 wrote: >> >> If you see that message then your machine/os/combo is having some problems >> keeping up. >> It's not the timer missing anything its the monotonic clock detecting a 1 >> second or more differential from what its next prediction for the time >> should be. The best way to trigger this would be to suspend FS with >> control-z or attach to it with gdb blocking the entire process, that 1ms >> thread would have to miss 1000 iterations to trigger that warning. >> >> Btw, that error message is at line 471 not 473 so you are using modified >> code. >> >> Its possible your box has a bad monotonic timer, you can set >> >> >> >> under in switch.conf.xml >> >> We are now starting to guess you are using some small embedded type >> platform >> perhaps? >> I've run FS even on a nokia n810 and never caused that message to fire. >> >> if 1 call can interrupt the cpu enough to cause noticeable issues you >> might >> want to consider running the process at a >> greater priority by using the -hp command line arg or at least nice it >> >> Why don't you tell us the whole story about what OS/platform you are using >> here rather that form conjectures about what is wrong with our code that >> thousands of people are happy with. >> >> >> >> >> >> >> >> On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: >> >>> >>> Btw, I have these popping up in my logs from time to time: >>> >>> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >>> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >>> Detected! Syncing Clock >>> >>> In this case an incoming call rang to both FS and Asterisk, Asterisk >>> picked >>> up, but the surge of activity made FS timer thread miss a beat or two. >>> >>> >>> eaf wrote: >>>> >>>> Oh, it's not just one timer thread... Why, why is sql_thread keeps on >>>> checking for messages every millisecond? Couldn't there be some >>> signalling >>>> implemented that will make the thread suspend on condition variable or >>> a >>>> socket/pipe in between? >>>> >>>> #0 do_sleep (t=1000) at src/switch_time.c:109 >>>> #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) >>> at >>>> src/switch_core_sqldb.c:783 >>>> >>>> Why does this sofia_profile_worker_thread keeps on looping checking for >>>> the queue? Have a semaphore! >>>> >>>> #0 do_sleep (t=1000) at src/switch_time.c:109 >>>> #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >>>> obj=0x80f2490) at sofia.c:978 >>>> >>>> Nothing's happening on the box, but there are three threads that >>> pretend >>>> to be actively busy with smth. Others at least sleep for hundreds of >>>> milliseconds, not for one. >>>> >>>> And there is even infrastructure present to do blocking pops: i.e. why >>>> couldn't sqldb thread do queue_pop() instead of queue_trypop() >>> intermixed >>>> with 1ms sleeps? This looping is such a waste... >>>> >>>> >>>> eaf wrote: >>>>> >>>>> As I see it, switch_cond_next() currently is just a do_sleep(1000). >>> Yes, >>>>> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >>>>> overrides that. >>>>> >>>>> Yeah, there is a global timestamp... It's easy to workaround that for >>> RTP >>>>> who calls switch_micro_time_now()... But if somebody accesses >>>>> runtime.timestamp directly, it's gonna be tough to grep for that. If >>> only >>>>> this was C++... >>>>> >>>>> I'll play around. Never liked polling too much. Never could've guessed >>>>> that polling could be so useful for scalability ;) My naive >>>>> implementation would've pulled timestamp via system calls and would've >>>>> done sleeping by passing exact interval to select() instead of syncing >>>>> with a pacing thread. Which would be dead-quiet at idle time, but, of >>>>> course, would stop scaling at some point due to excessive number of >>>>> system calls. >>>>> >>>>> Thanks. >>>>> >>>>> >>>>> Michael Jerris wrote: >>>>>> >>>>>> In short. No, you can not for many reasons. The milisecond tic is >>>>>> used throughout the code even when there is not any calls up. You >>> can >>>>>> grep for switch_cond_next if you would like to see where but it is >>>>>> required to keep our global timestamp and for pacing the scheduler >>>>>> among other services that run all the time. >>>>>> >>>>>> Mike >>>>>> >>>>>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >>>>>> >>>>>>> >>>>>>> Can I reduce resolution of that timer thread 10 times? I mean, I >>>>>>> glanced >>>>>>> through the code, and see that among others (are there others?) RTP >>>>>>> and IVR >>>>>>> set up their timers that are subsequently managed by this thread. >>>>>>> RTP timers >>>>>>> should be eliminated by that setting you've suggested. IVR timers >>>>>>> are set at >>>>>>> 20ms... So, if the thread is set to wake up every 10ms instead of >>>>>>> 1ms it >>>>>>> should be able to wake up those IVR timers just fine. Right? >>>>>>> >>>>>>> That's a cool design to have one dedicated thread that maintains >>>>>>> accurate >>>>>>> timing and then broadcasts via condition variables to hundreds of >>>>>>> other >>>>>>> threads events that they can register for. I'm sure it's one of the >>>>>>> reasons >>>>>>> why FS scales so much better than Asterisk. But for poor low-end >>>>>>> setups that >>>>>>> sit in the closet, eat only 6W of power and hardly ever run more >>>>>>> than two >>>>>>> calls at the same time, can I hack it somehow to be more UNIX- >>>>>>> friendly? I.e. >>>>>>> make it stuck in select() or recv() when there is nothing to do, >>> call >>>>>>> clock_gettime() right from the thread that wants and when it wants >>>>>>> to know >>>>>>> current time? >>>>>>> >>>>>>> Say, what if that thread is made to suspend on a condition variable >>>>>>> in case >>>>>>> if there are no timers registered in TIMER_MATRIX? Then, if some >>> other >>>>>>> thread comes up and adds its timer into the matrix, it could wake up >>>>>>> the >>>>>>> timer thread and enjoy accurate timing as needed, on demand? And in- >>>>>>> between >>>>>>> the calls, when there is no RTP or IVR, it will all go silent? I >>> mean, >>>>>>> sitting on a wait queue in the kernel is way better than go back and >>>>>>> forth >>>>>>> incrementing counters that nobody even needs at the moment? >>>>>>> >>>>>>> >>>>>>> Anthony Minessale-2 wrote: >>>>>>>> >>>>>>>> idle is a 4 letter word to a realtime application. >>>>>>>> >>>>>>>> The core keeps a single high-priority thread to keep 1ms timing and >>>>>>>> expands >>>>>>>> that broadcasting >>>>>>>> to hundreds or thousand of threads who need accurate timing. >>>>>>>> >>>>>>>> Your choppy audio is caused by linksys lying about the packet len >>>>>>>> that >>>>>>>> it's >>>>>>>> using and we set our timer >>>>>>>> to the wrong speed. >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> View this message in context: >>>>>>> >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>>>>> users >>>>>>> http://www.freeswitch.org >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Dec 3 09:57:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 09:57:53 -0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B17F95F.2000108@greatiam.com> References: <4B17F95F.2000108@greatiam.com> Message-ID: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah wrote: > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > Did you execute "reloadxml" from the fs cli before trying to connect with 2319? Also I'm assuming that "2319.xm" is a typo and you actually created "2319.xml" in the default/directory subdir. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/27520206/attachment.html From anthony.minessale at gmail.com Thu Dec 3 10:10:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 12:10:05 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> What about the things I spent time suggesting in my last email? Did you try them because I was actually curious if they made any impact. On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 > chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made > friendlier > to FS ;) > > > Anthony Minessale-2 wrote: > > > > If you see that message then your machine/os/combo is having some > problems > > keeping up. > > It's not the timer missing anything its the monotonic clock detecting a 1 > > second or more differential from what its next prediction for the time > > should be. The best way to trigger this would be to suspend FS with > > control-z or attach to it with gdb blocking the entire process, that 1ms > > thread would have to miss 1000 iterations to trigger that warning. > > > > Btw, that error message is at line 471 not 473 so you are using modified > > code. > > > > Its possible your box has a bad monotonic timer, you can set > > > > > > > > under in switch.conf.xml > > > > We are now starting to guess you are using some small embedded type > > platform > > perhaps? > > I've run FS even on a nokia n810 and never caused that message to fire. > > > > if 1 call can interrupt the cpu enough to cause noticeable issues you > > might > > want to consider running the process at a > > greater priority by using the -hp command line arg or at least nice it > > > > Why don't you tell us the whole story about what OS/platform you are > using > > here rather that form conjectures about what is wrong with our code that > > thousands of people are happy with. > > > > > > > > > > > > > > > > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > > > >> > >> Btw, I have these popping up in my logs from time to time: > >> > >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > >> Detected! Syncing Clock > >> > >> In this case an incoming call rang to both FS and Asterisk, Asterisk > >> picked > >> up, but the surge of activity made FS timer thread miss a beat or two. > >> > >> > >> eaf wrote: > >> > > >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > >> > checking for messages every millisecond? Couldn't there be some > >> signalling > >> > implemented that will make the thread suspend on condition variable or > >> a > >> > socket/pipe in between? > >> > > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) > >> at > >> > src/switch_core_sqldb.c:783 > >> > > >> > Why does this sofia_profile_worker_thread keeps on looping checking > for > >> > the queue? Have a semaphore! > >> > > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > >> > obj=0x80f2490) at sofia.c:978 > >> > > >> > Nothing's happening on the box, but there are three threads that > >> pretend > >> > to be actively busy with smth. Others at least sleep for hundreds of > >> > milliseconds, not for one. > >> > > >> > And there is even infrastructure present to do blocking pops: i.e. why > >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() > >> intermixed > >> > with 1ms sleeps? This looping is such a waste... > >> > > >> > > >> > eaf wrote: > >> >> > >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). > >> Yes, > >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> >> overrides that. > >> >> > >> >> Yeah, there is a global timestamp... It's easy to workaround that for > >> RTP > >> >> who calls switch_micro_time_now()... But if somebody accesses > >> >> runtime.timestamp directly, it's gonna be tough to grep for that. If > >> only > >> >> this was C++... > >> >> > >> >> I'll play around. Never liked polling too much. Never could've > guessed > >> >> that polling could be so useful for scalability ;) My naive > >> >> implementation would've pulled timestamp via system calls and > would've > >> >> done sleeping by passing exact interval to select() instead of > syncing > >> >> with a pacing thread. Which would be dead-quiet at idle time, but, of > >> >> course, would stop scaling at some point due to excessive number of > >> >> system calls. > >> >> > >> >> Thanks. > >> >> > >> >> > >> >> Michael Jerris wrote: > >> >>> > >> >>> In short. No, you can not for many reasons. The milisecond tic is > >> >>> used throughout the code even when there is not any calls up. You > >> can > >> >>> grep for switch_cond_next if you would like to see where but it is > >> >>> required to keep our global timestamp and for pacing the scheduler > >> >>> among other services that run all the time. > >> >>> > >> >>> Mike > >> >>> > >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >>> > >> >>>> > >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >> >>>> glanced > >> >>>> through the code, and see that among others (are there others?) RTP > >> >>>> and IVR > >> >>>> set up their timers that are subsequently managed by this thread. > >> >>>> RTP timers > >> >>>> should be eliminated by that setting you've suggested. IVR timers > >> >>>> are set at > >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of > >> >>>> 1ms it > >> >>>> should be able to wake up those IVR timers just fine. Right? > >> >>>> > >> >>>> That's a cool design to have one dedicated thread that maintains > >> >>>> accurate > >> >>>> timing and then broadcasts via condition variables to hundreds of > >> >>>> other > >> >>>> threads events that they can register for. I'm sure it's one of the > >> >>>> reasons > >> >>>> why FS scales so much better than Asterisk. But for poor low-end > >> >>>> setups that > >> >>>> sit in the closet, eat only 6W of power and hardly ever run more > >> >>>> than two > >> >>>> calls at the same time, can I hack it somehow to be more UNIX- > >> >>>> friendly? I.e. > >> >>>> make it stuck in select() or recv() when there is nothing to do, > >> call > >> >>>> clock_gettime() right from the thread that wants and when it wants > >> >>>> to know > >> >>>> current time? > >> >>>> > >> >>>> Say, what if that thread is made to suspend on a condition variable > >> >>>> in case > >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some > >> other > >> >>>> thread comes up and adds its timer into the matrix, it could wake > up > >> >>>> the > >> >>>> timer thread and enjoy accurate timing as needed, on demand? And > in- > >> >>>> between > >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I > >> mean, > >> >>>> sitting on a wait queue in the kernel is way better than go back > and > >> >>>> forth > >> >>>> incrementing counters that nobody even needs at the moment? > >> >>>> > >> >>>> > >> >>>> Anthony Minessale-2 wrote: > >> >>>>> > >> >>>>> idle is a 4 letter word to a realtime application. > >> >>>>> > >> >>>>> The core keeps a single high-priority thread to keep 1ms timing > and > >> >>>>> expands > >> >>>>> that broadcasting > >> >>>>> to hundreds or thousand of threads who need accurate timing. > >> >>>>> > >> >>>>> Your choppy audio is caused by linksys lying about the packet len > >> >>>>> that > >> >>>>> it's > >> >>>>> using and we set our timer > >> >>>>> to the wrong speed. > >> >>>>> > >> >>>>> > >> >>>> > >> >>>> -- > >> >>>> View this message in context: > >> >>>> > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >>>> > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch- > >> >>>> users > >> >>>> http://www.freeswitch.org > >> >>> > >> >>> _______________________________________________ > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >>> > >> >> > >> >> > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/d9ff3e1b/attachment-0001.html From freeswitch-users-list at metik.com Thu Dec 3 10:17:43 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 03 Dec 2009 13:17:43 -0500 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: References: Message-ID: <4B1800C7.7010800@metik.com> Yehavi, There are a few variations of transmitting this information... If you have already enabled a supplemental isdn service profile, try adding the following to the PRI you are using: (config-if)#isdn outgoing ie facility (config-if)#iisdn outgoing ie extended-facility (config-if)#isdn outgoing display-ie (config-if)#isdn outgoing ie caller-number (config-if)#isdn outgoing ie called-number -metik Yehavi Bourvine wrote: > Hello, > > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On > the PRI there is a Nortel with Q.Sig. After a lot of configuration > trials I've managed to set it to send back the connected name over the > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the > connected name and then the Cisco adds it as a Remote-Party-ID). > However, I did not save it and a power outage cleared this config. In > my age I don't remember what I've done... > > Anyone knows the correct config? > > Thanks! __Yehavi: > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shiyanov at gmail.com Thu Dec 3 10:20:12 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 3 Dec 2009 21:20:12 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> Message-ID: I've sent deep-breath message to the dev list. Just-in-case, here is a "cross-post": Hi there! This message is a forward from user-mail-list. I'm trying to fix such a problem: FreSwithch compiled from SVN-trunk, date = 11/02/2009. What is need: connect two users, initially one is on the home-grown java-based IVR and other party is off hook. What is done/got: User1 is on the java application, it represents simple IVR system, and the most used FS API operation is "streamFile". User2 is off hook. next: (mod_socket) create_uuid bgapi originate {origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park() uuid_bridge uuid_User1 uuid_User2 FS log is here: http://pastebin.freeswitch.org/11380 Thank you much for any help, Artem On Wed, Dec 2, 2009 at 10:24 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you should be working on SVN trunk if you are doing development, we are so > far forward from 1.0.4 we can't do debugging very easily. > > I don't know all of the details of what you are trying to do but you are > hitting some race conditions because of the async nature of the socket > connection and the way you are using it. > > > > > On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov wrote: > >> I'm back again with the same issue. >> Now it is became worse: it reproduces occasionally. >> [FS version is 1.04, test_load = 2 active calls] >> >> I've got 2 logs: successful and not. >> Here is a bad_case: >> >> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute >> java(/usr/local/freeswitch/scripts/fs2agi.jar >> org.starpound.fs2agi.Translator >> ${agi_url}) >> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run >> INFO: *************************************************** >> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run >> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d >> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI >> application >> agi://localhost:4573/hello.agi?callId=929 >> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! >> >> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] >> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready >> sofia/internal/2001! >> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel >> [sofia/internal/2001] has >> been answered >> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: >> java.lang.Exception: Internal FreeSwitch failure while streamming file, >> see >> FreeSwitch logs for details >> at >> >> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) >> at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) >> at org.starpound.fs2agi.Translator.run(Translator.java:56) >> at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) >> at >> >> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) >> at >> >> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) >> at java.lang.reflect.Method.invoke(Method.java:597) >> at org.freeswitch.Launcher.launch(Launcher.java:80) >> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup >> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application >> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for >> details. >> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup >> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] >> [DESTINATION_OUT_OF_ORDER] >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 >> (sofia/external/6786081291 at 66.19.38.143) Ended >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close >> Channel >> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 >> (sofia/internal/2001) Ended >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close >> Channel >> sofia/internal/2001 [CS_DESTROY] >> >> >> >> Message >> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: >> ..." >> is sent from my app upon the onHangup().` >> >> And here is a good_case: >> >> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute >> java(/usr/local/freeswitch/scripts/fs2agi.jar >> org.starpound.fs2agi.Translator >> ${agi_url}) >> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run >> INFO: *************************************************** >> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run >> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for >> session >> 7c37369b-ffb2-4436-9288-a640047d0e5e >> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI >> application >> agi://localhost:4573/hello.agi?callId=932 >> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! >> >> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] >> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready >> sofia/internal/2001! >> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel >> [sofia/internal/2001] has >> been answered >> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for >> session >> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: >> java.lang.Exception: Internal FreeSwitch failure while streamming file, >> see >> FreeSwitch logs for details >> at >> >> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) >> at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) >> at org.starpound.fs2agi.Translator.run(Translator.java:56) >> at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) >> at >> >> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) >> at >> >> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) >> at java.lang.reflect.Method.invoke(Method.java:597) >> at org.freeswitch.Launcher.launch(Launcher.java:80) >> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application >> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for >> details. >> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port >> from >> 172.26.10.39:26402 to 91.190.120.190:26402 >> >> >> >> Suggestions? >> >> >> >> >> >> >> >> >> >> >> >> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: >> >>> Anthony, >>> >>> >>As soon as you call uuid_bridge you are transferring both legs of the >>> call to bridge to each other. >>> >>This means your java app must exit so the channels can connect to each >>> other. >>> >>> I didn't know that. Now my java app is exiting upon the onHangup() call >>> so everything has become "ok". Thank you much. >>> I'll add note to the wiki about this issue. >>> >>> Artem >>> >>> >>> >>> >>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Your "annoying behaviour" is the exact behavior you should be getting >>>> considering what you told FS to do. >>>> >>>> As soon as you call uuid_bridge you are transferring both legs of the >>>> call to bridge to each other. >>>> This means your java app must exit so the channels can connect to each >>>> other. >>>> >>>> remember that you hangup hook can be called when the channel is >>>> transferred not only when it hangs up. >>>> you have to test which is happening based on the input to your callback. >>>> >>>> >>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >>>> >>>>> Hi there! >>>>> >>>>> I've got annoying FS behavior: >>>>> There are 2 channels executing the same Java application (application >>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>>>> channels are killed. Here is a log from FS console: >>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>>>> CS_HIBERNATE >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>>> called >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>>>> playing file >>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>>>> playing file >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send >>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK] >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>>>> CS_EXECUTE -> CS_HIBERNATE >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>>> called >>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>>> >>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>>>> switch_core_session.c:933 Send signal >>>>> sofia/internal/1001 at master.agent.starpoundtec >>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send >>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK] >>>>> >>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send >>>>> signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>>>> >>>>> (FS version is 1.0.4) >>>>> >>>>> Any thoughts? >>>>> >>>>> >>>>> Artem >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/90a40da6/attachment-0001.html From dlaperle at rsslex.com Thu Dec 3 10:29:13 2009 From: dlaperle at rsslex.com (David Laperle) Date: Thu, 03 Dec 2009 13:29:13 -0500 Subject: [Freeswitch-users] Dialplan behavior Message-ID: <1259864953.1978.12.camel@dlaplap> Hi guys, i have a weird problem with my dialplans. For the moment, i have only 2 ??usable?? extensions. They were working #1 yesterday, but this morning i realize i forgot to compile mod_python, so i go back into my source folder and modify the modules.conf to uncomment mod_python, did a make and make install (i did a backup of my conf folder before)! The make and make install worked flawlessly. Then i put back my bkp of conf directory. I restarted the freeswitch service, created my python test dialplan and entered into cli to see what's gonna happen! To my surprise, the call didn't processed to the extension i was dialing. i tried all the other extensions i had, they were all not working!!!! After that i realized that the .xml in freeswitch/dialplan/default/ weren't imported into configuration at startup ... I have read all the documentation about difference between public and default dialplan and i understand them correctly, in public if i include all default folder, it's working again (i can reach all my extensions in default. My extensions are in the correct user_context ... i did nothing since yesterday other than a make && make install after enabling python ... Any other user have an idea why the default/*.xml aren't processed automatically? What could i have done wrong so they are no longer processed? Thanks a lot, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. Bien que ce courriel ait ?t? trait? contre les virus, il est de la responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot?g?s par le secret professionnel. Si vous d?sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier ?lectronique ordinaire non s?curis?, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/1d1b11fa/attachment.html From abeka at greatiam.com Thu Dec 3 10:34:25 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:34:25 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B1804B1.2060104@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From abeka at greatiam.com Thu Dec 3 10:34:45 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:34:45 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B1804C5.5070302@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From abeka at greatiam.com Thu Dec 3 10:36:21 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:36:21 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B180525.7060702@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From erandr-junk at usa.net Thu Dec 3 10:43:40 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 10:43:40 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> Message-ID: <26630994.post@talk.nabble.com> You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I thought I responded back. Perhaps it didn't make through though, as I just emailed back to the list instead of using nabble.com... Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went w/o any effect either, but disabling RTP timers did the trick. I don't have the original "choppy sound with PCMU" problem any more, thanks a lot for the quick turnaround on that question. But your suggestions made me look, into logs, strace, code, etc, so now I'm just checking on how to quiet down those busy loops a little and how to get rid of periodic CRIT messages about Virtual Machine Migration. Anthony Minessale-2 wrote: > > What about the things I spent time suggesting in my last email? > Did you try them because I was actually curious if they made any impact. > > > On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > >> >> I'm sorry if I sounded that way. Did mean to. :) >> >> Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 >> chip >> and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm >> >> Line offset difference is due to some minor logging changes I made to see >> who's allocating timers and how often. This way I found MOH streaming and >> that RTP still allocates timers even when it's set to none in the >> profile. >> >> I feel that this platform turned out to be underpowered for FS because it >> cannot meet its scheduling expectations. I guess, some degree of kernel >> tweaking or setting priorities will fix that. Meanwhile I just got rid of >> the SQLDB 1ms thread via -nosql command line option, split sofia worker >> 1ms >> thread in two (one blocked and waiting for new commands in the SQL queue, >> the other one checking registrations and gateways with 1sec interval), >> and >> don't know yet what to do about the timer thread. >> >> Again, I apologize for stupid or accusing questions, I'm just trying to >> see >> how FS can be made friendlier to this board. Or the board be made >> friendlier >> to FS ;) >> >> >> Anthony Minessale-2 wrote: >> > >> > If you see that message then your machine/os/combo is having some >> problems >> > keeping up. >> > It's not the timer missing anything its the monotonic clock detecting a >> 1 >> > second or more differential from what its next prediction for the time >> > should be. The best way to trigger this would be to suspend FS with >> > control-z or attach to it with gdb blocking the entire process, that >> 1ms >> > thread would have to miss 1000 iterations to trigger that warning. >> > >> > Btw, that error message is at line 471 not 473 so you are using >> modified >> > code. >> > >> > Its possible your box has a bad monotonic timer, you can set >> > >> > >> > >> > under in switch.conf.xml >> > >> > We are now starting to guess you are using some small embedded type >> > platform >> > perhaps? >> > I've run FS even on a nokia n810 and never caused that message to fire. >> > >> > if 1 call can interrupt the cpu enough to cause noticeable issues you >> > might >> > want to consider running the process at a >> > greater priority by using the -hp command line arg or at least nice it >> > >> > Why don't you tell us the whole story about what OS/platform you are >> using >> > here rather that form conjectures about what is wrong with our code >> that >> > thousands of people are happy with. >> > >> > >> > >> > >> > >> > >> > >> > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: >> > >> >> >> >> Btw, I have these popping up in my logs from time to time: >> >> >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >> >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >> >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >> >> Detected! Syncing Clock >> >> >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk >> >> picked >> >> up, but the surge of activity made FS timer thread miss a beat or two. >> >> >> >> >> >> eaf wrote: >> >> > >> >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps >> on >> >> > checking for messages every millisecond? Couldn't there be some >> >> signalling >> >> > implemented that will make the thread suspend on condition variable >> or >> >> a >> >> > socket/pipe in between? >> >> > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, >> obj=0x0) >> >> at >> >> > src/switch_core_sqldb.c:783 >> >> > >> >> > Why does this sofia_profile_worker_thread keeps on looping checking >> for >> >> > the queue? Have a semaphore! >> >> > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >> >> > obj=0x80f2490) at sofia.c:978 >> >> > >> >> > Nothing's happening on the box, but there are three threads that >> >> pretend >> >> > to be actively busy with smth. Others at least sleep for hundreds of >> >> > milliseconds, not for one. >> >> > >> >> > And there is even infrastructure present to do blocking pops: i.e. >> why >> >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() >> >> intermixed >> >> > with 1ms sleeps? This looping is such a waste... >> >> > >> >> > >> >> > eaf wrote: >> >> >> >> >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). >> >> Yes, >> >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> >> >> overrides that. >> >> >> >> >> >> Yeah, there is a global timestamp... It's easy to workaround that >> for >> >> RTP >> >> >> who calls switch_micro_time_now()... But if somebody accesses >> >> >> runtime.timestamp directly, it's gonna be tough to grep for that. >> If >> >> only >> >> >> this was C++... >> >> >> >> >> >> I'll play around. Never liked polling too much. Never could've >> guessed >> >> >> that polling could be so useful for scalability ;) My naive >> >> >> implementation would've pulled timestamp via system calls and >> would've >> >> >> done sleeping by passing exact interval to select() instead of >> syncing >> >> >> with a pacing thread. Which would be dead-quiet at idle time, but, >> of >> >> >> course, would stop scaling at some point due to excessive number of >> >> >> system calls. >> >> >> >> >> >> Thanks. >> >> >> >> >> >> >> >> >> Michael Jerris wrote: >> >> >>> >> >> >>> In short. No, you can not for many reasons. The milisecond tic is >> >> >>> used throughout the code even when there is not any calls up. You >> >> can >> >> >>> grep for switch_cond_next if you would like to see where but it is >> >> >>> required to keep our global timestamp and for pacing the scheduler >> >> >>> among other services that run all the time. >> >> >>> >> >> >>> Mike >> >> >>> >> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >> >>> >> >> >>>> >> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >> >> >>>> glanced >> >> >>>> through the code, and see that among others (are there others?) >> RTP >> >> >>>> and IVR >> >> >>>> set up their timers that are subsequently managed by this thread. >> >> >>>> RTP timers >> >> >>>> should be eliminated by that setting you've suggested. IVR timers >> >> >>>> are set at >> >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >> >> >>>> 1ms it >> >> >>>> should be able to wake up those IVR timers just fine. Right? >> >> >>>> >> >> >>>> That's a cool design to have one dedicated thread that maintains >> >> >>>> accurate >> >> >>>> timing and then broadcasts via condition variables to hundreds of >> >> >>>> other >> >> >>>> threads events that they can register for. I'm sure it's one of >> the >> >> >>>> reasons >> >> >>>> why FS scales so much better than Asterisk. But for poor low-end >> >> >>>> setups that >> >> >>>> sit in the closet, eat only 6W of power and hardly ever run more >> >> >>>> than two >> >> >>>> calls at the same time, can I hack it somehow to be more UNIX- >> >> >>>> friendly? I.e. >> >> >>>> make it stuck in select() or recv() when there is nothing to do, >> >> call >> >> >>>> clock_gettime() right from the thread that wants and when it >> wants >> >> >>>> to know >> >> >>>> current time? >> >> >>>> >> >> >>>> Say, what if that thread is made to suspend on a condition >> variable >> >> >>>> in case >> >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some >> >> other >> >> >>>> thread comes up and adds its timer into the matrix, it could wake >> up >> >> >>>> the >> >> >>>> timer thread and enjoy accurate timing as needed, on demand? And >> in- >> >> >>>> between >> >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I >> >> mean, >> >> >>>> sitting on a wait queue in the kernel is way better than go back >> and >> >> >>>> forth >> >> >>>> incrementing counters that nobody even needs at the moment? >> >> >>>> >> >> >>>> >> >> >>>> Anthony Minessale-2 wrote: >> >> >>>>> >> >> >>>>> idle is a 4 letter word to a realtime application. >> >> >>>>> >> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing >> and >> >> >>>>> expands >> >> >>>>> that broadcasting >> >> >>>>> to hundreds or thousand of threads who need accurate timing. >> >> >>>>> >> >> >>>>> Your choppy audio is caused by linksys lying about the packet >> len >> >> >>>>> that >> >> >>>>> it's >> >> >>>>> using and we set our timer >> >> >>>>> to the wrong speed. >> >> >>>>> >> >> >>>>> >> >> >>>> >> >> >>>> -- >> >> >>>> View this message in context: >> >> >>>> >> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> >> >>>> Sent from the Freeswitch-users mailing list archive at >> Nabble.com. >> >> >>>> >> >> >>>> >> >> >>>> _______________________________________________ >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch- >> >> >>>> users >> >> >>>> http://www.freeswitch.org >> >> >>> >> >> >>> _______________________________________________ >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >>> >> >> >> >> >> >> >> >> > >> >> > >> >> >> >> -- >> >> View this message in context: >> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26630994.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mustafa.pk at gmail.com Thu Dec 3 10:46:44 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Thu, 3 Dec 2009 23:46:44 +0500 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <1259864953.1978.12.camel@dlaplap> References: <1259864953.1978.12.camel@dlaplap> Message-ID: <8213d6070912031046y6cd97f0cj38c9d392f92dfdda@mail.gmail.com> other than configuration/syntax problem it could be a simple character/file encoding problem or may be improper file permissions! On Thu, Dec 3, 2009 at 11:29 PM, David Laperle wrote: > Hi guys, > > i have a weird problem with my dialplans. For the moment, i have only 2 > ?usable? extensions. They were working #1 yesterday, but this morning i > realize i forgot to compile mod_python, so i go back into my source folder > and modify the modules.conf to uncomment mod_python, did a make and make > install (i did a backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test dialplan and > entered into cli to see what's gonna happen! To my surprise, the call didn't > processed to the extension i was dialing. > > i tried all the other extensions i had, they were all not working!!!! > > After that i realized that the .xml in freeswitch/dialplan/default/ weren't > imported into configuration at startup ... > > I have read all the documentation about difference between public and > default dialplan and i understand them correctly, in public if i include all > default folder, it's working again (i can reach all my extensions in > default. > > My extensions are in the correct user_context ... i did nothing since > yesterday other than a make && make install after enabling python ... > > Any other user have an idea why the default/*.xml aren't processed > automatically? What could i have done wrong so they are no longer processed? > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > *Robinson Sheppard Shapiro *s.e.n.c.r.l/LLP > Avocats / Barristers & Solicitors > 4600 - 800 Place Victoria > Montr?al Qc H4Z 1H6 > T (514) 878-2631 F (514) 878-1865 > www.rsslex.com et/and www.rsscanadaimmigration.com > > > > > * > ------------------------------ > **http://www.rsslex.com** * > > *AVIS:* Ce courriel privil?gi? et confidentiel est destin? ? la seule > personne ou entit? ? laquelle il est adress?. Pour toute autre personne, > toute action prise en rapport ? ce courriel ainsi que toute lecture, > reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble > de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? > recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. > Bien que ce courriel ait ?t? trait? contre les virus, il est de la > responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos > communications avec vous peuvent contenir des renseignements confidentiels > ou prot?g?s par le secret professionnel. Si vous d?sirez que nous > communiquions avec vous par un autre moyen de transmission que le courrier > ?lectronique ordinaire non s?curis?, veuillez nous en aviser. > > *NOTICE:* This privileged and confidential email is intended only for the > individual or entity to whom it is addressed. With regard to all others, any > action related with this email as well as any reading, reproduction, > transmission and/or dissemination in whole or in part of the information > included in this email is prohibited. If you are not the addressee, > immediately return the email to sender prior to destroying all copies. Even > if this email is believed to be free from any virus, it is the > responsibility of the recipient to make sure that it is virus exempt. Our > communications to you may contain confidential information or information > protected under solicitor-client privilege. Please advise if you wish us to > use a mode of communication other than regular, unsecured e-mail in our > communications with you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/48c9e302/attachment.html From asannucci at gmail.com Thu Dec 3 10:47:16 2009 From: asannucci at gmail.com (bakko) Date: Thu, 3 Dec 2009 13:47:16 -0500 Subject: [Freeswitch-users] can't register Inphonex In-Reply-To: <005a01ca73c9$bc2dcf60$34896e20$@ca> References: <005a01ca73c9$bc2dcf60$34896e20$@ca> Message-ID: <3A8174E906FB45CDA04B78C41ED21A88@voztovoice> >From de console: sofia profile external siptrace on or with ngrep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/ad12a5b6/attachment.html From davis.erwin at gmail.com Thu Dec 3 11:30:49 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Thu, 3 Dec 2009 14:30:49 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> Message-ID: Hi, Anthony and Mike, With the latest version from SVN, I was able to remove the warning "sample rate not matching". But the remote RTP port was still changed after after playing the vm greeting. See below, 2009-12-03 13:44:46.901216 [INFO] switch_rtp.c:1975 Auto Changing port from XXX.YYY.ZZZ.39:10002 to XXX.YYY.ZZZ.39:3335 Any clue? I looked at the source code in switch_rtp.c:1975, it shows that if rtp_session->autoadj_tally >= 10, then a rtp port change will happen. Any idea about autoadj_tally and what cause the increase of autoadj_tally ? Thanks, On 12/2/09, Erwin Davis wrote: > > Hi, Anthony and Mike, > > Thanks for your reply. The problem still exists even after I ran "make > hd-sounds install". > I will try the latest version from the SVN to see if the problem will go > away. I will let you know. > Thanks folks, > > Regards, > > On 12/2/09, Michael Collins wrote: > >> >> >> On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: >> >>> Hi, Anthony, >>> >>> Thanks for your reply. >>> >>> When I type the command below, I got the error, >>> Unknown target hd-sound-install >>> make[1]: *** [hd-sound-install] Error 1 >>> make: *** [hd-sound-install] Error 2 >>> >>> I found out that under >>> /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, >>> 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should >>> explain why at first FS played in right sample rate. But after playing >>> serveral time, FS complained about sample rate not matching. Any clue? >>> Thanks, >>> >>> >> Erwin, >> >> As Tony said you've actually got a pretty old installation. If this is in >> production then I would recommend getting a sandbox machine, install trunk >> using the quick-and-dirty install, and then update the default config to you >> specific configuration. Test to make sure it works before you put it into >> production. :) >> >> Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run >> into any issues that require more real-time conversation. >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/447f2cfa/attachment.html From yehavi.bourvine at gmail.com Thu Dec 3 11:31:57 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 21:31:57 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: <4B1800C7.7010800@metik.com> References: <4B1800C7.7010800@metik.com> Message-ID: Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back to ISDN the name of the destination, but not the other way around... Thanks! __Yehavi: 2009/12/3 Metik > Yehavi, > > There are a few variations of transmitting this information... If you > have already enabled a supplemental isdn service profile, try adding the > following to the PRI you are using: > > (config-if)#isdn outgoing ie facility > (config-if)#iisdn outgoing ie extended-facility > (config-if)#isdn outgoing display-ie > (config-if)#isdn outgoing ie caller-number > (config-if)#isdn outgoing ie called-number > > -metik > > Yehavi Bourvine wrote: > > Hello, > > > > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On > > the PRI there is a Nortel with Q.Sig. After a lot of configuration > > trials I've managed to set it to send back the connected name over the > > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the > > connected name and then the Cisco adds it as a Remote-Party-ID). > > However, I did not save it and a power outage cleared this config. In > > my age I don't remember what I've done... > > > > Anyone knows the correct config? > > > > Thanks! __Yehavi: > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/998e13cf/attachment-0001.html From msc at freeswitch.org Thu Dec 3 11:51:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 11:51:09 -0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B1804B1.2060104@greatiam.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> <4B1804B1.2060104@greatiam.com> Message-ID: <87f2f3b90912031151kf2a6843w1b48cf36330a9252@mail.gmail.com> On Thu, Dec 3, 2009 at 10:34 AM, Samuel Abekah-Mensah wrote: > Hi > > Sorry .xm is a typo. I actually shut down the server and restarted. The > log says I need to create a domain of aaa.bbb.ccc.ddd (which is the > server IP address ) and then put the user in that domain. Isn't the > default domain that of the server FS is running on ? > 2319.xml is in /usr/local/freeswitch/conf/directory/default/ > > Thanks for your time > > Okay, here's exactly what I did: cd /usr/local/freeswitch/conf/directory/default cp 1001.xml 2319.xml perl -pi -e 's/1001/2319/g' 2319.xml cat 2319.xml Then I logged into fs_cli, pressed F6 (which does "reloadxml") and then I set up my x-lite: Display Name: Test User name: 2319 Password: 1234 Authorization user name: 2319 Domain: 10.15.0.91 It registered just fine as can be seen by the output of "sofia status profile internal": Call-ID: MzRiOGI4NTA2YjA0ZTkzMDYwZjA3MTlkZGQ3ZjNhMjg. User: 2319 at 10.15.0.91 Contact: "Test" Agent: X-Lite release 1014k stamp 47051 Status: Registered(UDP)(unknown) EXP(2009-12-03 13:41:38) Host: freeswitch1.yt IP: 10.15.0.124 Port: 41680 Auth-User: 2319 Auth-Realm: 10.15.0.91 MWI-Account: 2319 at 10.15.0.91 So, most likely you've got an issue with the XML file itself or the configuration on your SIP device. Double check the username and auth username values. If need be delete your 2319.xml file and start over. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/cb357bb8/attachment.html From msc at freeswitch.org Thu Dec 3 11:55:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 11:55:23 -0800 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <1259864953.1978.12.camel@dlaplap> References: <1259864953.1978.12.camel@dlaplap> Message-ID: <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> On Thu, Dec 3, 2009 at 10:29 AM, David Laperle wrote: > Hi guys, > > i have a weird problem with my dialplans. For the moment, i have only 2 > ?usable? extensions. They were working #1 yesterday, but this morning i > realize i forgot to compile mod_python, so i go back into my source folder > and modify the modules.conf to uncomment mod_python, did a make and make > install (i did a backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test dialplan and > entered into cli to see what's gonna happen! To my surprise, the call didn't > processed to the extension i was dialing. > > i tried all the other extensions i had, they were all not working!!!! > > After that i realized that the .xml in freeswitch/dialplan/default/ weren't > imported into configuration at startup ... > > I have read all the documentation about difference between public and > default dialplan and i understand them correctly, in public if i include all > default folder, it's working again (i can reach all my extensions in > default. > > My extensions are in the correct user_context ... i did nothing since > yesterday other than a make && make install after enabling python ... > > Any other user have an idea why the default/*.xml aren't processed > automatically? What could i have done wrong so they are no longer processed? > > double-check for the existence of conf/dialplan/default.xml - I've seen on rare occasion where that file simple goes away for no apparent reason. Since I never change that file - and I recommend that you never change it either ;) - you can go to your FS source directory and issue "make samples" and it will re-create any missing default config files without overwriting you existing config files. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/6a7af16f/attachment.html From jbr at consiglia.dk Thu Dec 3 12:05:17 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Thu, 3 Dec 2009 21:05:17 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I am trying to rewrite all my javascript scripts into Lua scripts. I have run into the problem of core_db access. This can be achieved with Spidermonkey, but apparently not with Lua. I have tried to get the binary for Lua (using apt-get) but I get an error when I require the sqlite.so: undefined symbol: luaopen_luasql_sqlite, so I'm stuck. So what is a feasible way to manipulate the core database from Lua? I may mention that access to MySQL works perfectly from Lua. Regards Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/43f1b4ac/attachment.html From anthony.minessale at gmail.com Thu Dec 3 12:29:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 14:29:13 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912031229g7b6bf3cdo43b58c43cd2be6a8@mail.gmail.com> In latest trunk you can run the core db in your same mysql db. other than that we would need to create an object from our lua module similar to how it was done in js. On Thu, Dec 3, 2009 at 2:05 PM, Jon Bruel wrote: > I am trying to rewrite all my javascript scripts into Lua scripts. I have > run into the problem of core_db access. This can be achieved with > Spidermonkey, but apparently not with Lua. I have tried to get the binary > for Lua (using apt-get) but I get an error when I require the sqlite.so: > undefined symbol: luaopen_luasql_sqlite, so I?m stuck. So what is a feasible > way to manipulate the core database from Lua? > > I may mention that access to MySQL works perfectly from Lua. > > Regards Jon > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/5d1a37cd/attachment.html From timuckun at gmail.com Thu Dec 3 12:40:16 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 09:40:16 +1300 Subject: [Freeswitch-users] HA questions. In-Reply-To: <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> Message-ID: <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: > The easiest place to do this is at the point you send the calls to FreeSWITCH. ?How are the calls coming in? > >From an as of now unkown SIP trunk provider (we are still in negotiations with a couple of companies). From davis.erwin at gmail.com Thu Dec 3 12:41:00 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Thu, 3 Dec 2009 15:41:00 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: Message-ID: Hi, I solved this issue. the reason is because of the different port number between the the one in SDP and the one in real RTP stream. This is very nice feature. e On 12/2/09, Erwin Davis wrote: > > Hi, I got a weird issue when I dialed an extension and listen to a recorded > voice mail greeting message. > After playing a couple of time of the greeting, the FS printed the warning > of "sample rate not matching", then > send the audio to a different remote RTP port. See the log below, > > > 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message.wav] (en:en) > 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate > doesn't match > 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec > Activated > 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from > xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 > 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original > codec. > 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less > than minimum record length: 3, discarding it. > 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-too-small.wav] (en:en) > 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ > > > the original codec is wideband 16kHz Speex and the wireshark shows that the > FS used the same codec. I used FS 1.04 in fedora 8. > I have two questions here, > (1) why does FS report "Sample rate doesn't match"? is it a bug or > configuration issue? > (2) Why does FS change the RTP port ? how to fix it? > > Thanks, > > Regards, > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8b5bd90c/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 3 12:40:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 14:40:59 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26630994.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> Message-ID: <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> no, I mean the one after that that you must have completely skipped with a command line option to try and a param to set in the config. It somewhat annoys me for taking the time to compose it now. I wrote all of the code you are talking about myself and I was trying to give you some suggestions.... Well, actually, you did answer my question about the platform so you must have seen it..... The loops are not the cause of that migration message, something wrong with the hardware or the kernel is. Another guy just told you he does not see that problem on the same exact hardware. Even if you have a point about the sql threads, you could make a patch to slow them down but you cant slow down too much or you will not be able to handle 400 cps all asking to send updates to transactions in batches of thousands of sql stmts. Every line of that code is carefully designed so I don't know what else to tell you but to stop being so arrogant and re-read this thread for all the advice you have totally ignored. I started out trying to help you but I have a lot of work to do. I thoroughly explained it to you and you are choosing to ignore me so I guess I'm done. You can do whatever you want with your working copy, i'll see you in 3 or 4 years when you get up to speed with the rest of us........ On Thu, Dec 3, 2009 at 12:43 PM, eaf wrote: > > You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I > thought I responded back. Perhaps it didn't make through though, as I just > emailed back to the list instead of using nabble.com... > > Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went > w/o any effect either, but disabling RTP timers did the trick. I don't have > the original "choppy sound with PCMU" problem any more, thanks a lot for > the > quick turnaround on that question. > > But your suggestions made me look, into logs, strace, code, etc, so now I'm > just checking on how to quiet down those busy loops a little and how to get > rid of periodic CRIT messages about Virtual Machine Migration. > > > Anthony Minessale-2 wrote: > > > > What about the things I spent time suggesting in my last email? > > Did you try them because I was actually curious if they made any impact. > > > > > > On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > > > >> > >> I'm sorry if I sounded that way. Did mean to. :) > >> > >> Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 > >> chip > >> and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > >> > >> Line offset difference is due to some minor logging changes I made to > see > >> who's allocating timers and how often. This way I found MOH streaming > and > >> that RTP still allocates timers even when it's set to none in the > >> profile. > >> > >> I feel that this platform turned out to be underpowered for FS because > it > >> cannot meet its scheduling expectations. I guess, some degree of kernel > >> tweaking or setting priorities will fix that. Meanwhile I just got rid > of > >> the SQLDB 1ms thread via -nosql command line option, split sofia worker > >> 1ms > >> thread in two (one blocked and waiting for new commands in the SQL > queue, > >> the other one checking registrations and gateways with 1sec interval), > >> and > >> don't know yet what to do about the timer thread. > >> > >> Again, I apologize for stupid or accusing questions, I'm just trying to > >> see > >> how FS can be made friendlier to this board. Or the board be made > >> friendlier > >> to FS ;) > >> > >> > >> Anthony Minessale-2 wrote: > >> > > >> > If you see that message then your machine/os/combo is having some > >> problems > >> > keeping up. > >> > It's not the timer missing anything its the monotonic clock detecting > a > >> 1 > >> > second or more differential from what its next prediction for the time > >> > should be. The best way to trigger this would be to suspend FS with > >> > control-z or attach to it with gdb blocking the entire process, that > >> 1ms > >> > thread would have to miss 1000 iterations to trigger that warning. > >> > > >> > Btw, that error message is at line 471 not 473 so you are using > >> modified > >> > code. > >> > > >> > Its possible your box has a bad monotonic timer, you can set > >> > > >> > > >> > > >> > under in switch.conf.xml > >> > > >> > We are now starting to guess you are using some small embedded type > >> > platform > >> > perhaps? > >> > I've run FS even on a nokia n810 and never caused that message to > fire. > >> > > >> > if 1 call can interrupt the cpu enough to cause noticeable issues you > >> > might > >> > want to consider running the process at a > >> > greater priority by using the -hp command line arg or at least nice it > >> > > >> > Why don't you tell us the whole story about what OS/platform you are > >> using > >> > here rather that form conjectures about what is wrong with our code > >> that > >> > thousands of people are happy with. > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > >> > > >> >> > >> >> Btw, I have these popping up in my logs from time to time: > >> >> > >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > >> >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > >> >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > >> >> Detected! Syncing Clock > >> >> > >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk > >> >> picked > >> >> up, but the surge of activity made FS timer thread miss a beat or > two. > >> >> > >> >> > >> >> eaf wrote: > >> >> > > >> >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps > >> on > >> >> > checking for messages every millisecond? Couldn't there be some > >> >> signalling > >> >> > implemented that will make the thread suspend on condition variable > >> or > >> >> a > >> >> > socket/pipe in between? > >> >> > > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, > >> obj=0x0) > >> >> at > >> >> > src/switch_core_sqldb.c:783 > >> >> > > >> >> > Why does this sofia_profile_worker_thread keeps on looping checking > >> for > >> >> > the queue? Have a semaphore! > >> >> > > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> >> > #1 0xb73a4701 in sofia_profile_worker_thread_run > (thread=0x80f3a30, > >> >> > obj=0x80f2490) at sofia.c:978 > >> >> > > >> >> > Nothing's happening on the box, but there are three threads that > >> >> pretend > >> >> > to be actively busy with smth. Others at least sleep for hundreds > of > >> >> > milliseconds, not for one. > >> >> > > >> >> > And there is even infrastructure present to do blocking pops: i.e. > >> why > >> >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() > >> >> intermixed > >> >> > with 1ms sleeps? This looping is such a waste... > >> >> > > >> >> > > >> >> > eaf wrote: > >> >> >> > >> >> >> As I see it, switch_cond_next() currently is just a > do_sleep(1000). > >> >> Yes, > >> >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> >> >> overrides that. > >> >> >> > >> >> >> Yeah, there is a global timestamp... It's easy to workaround that > >> for > >> >> RTP > >> >> >> who calls switch_micro_time_now()... But if somebody accesses > >> >> >> runtime.timestamp directly, it's gonna be tough to grep for that. > >> If > >> >> only > >> >> >> this was C++... > >> >> >> > >> >> >> I'll play around. Never liked polling too much. Never could've > >> guessed > >> >> >> that polling could be so useful for scalability ;) My naive > >> >> >> implementation would've pulled timestamp via system calls and > >> would've > >> >> >> done sleeping by passing exact interval to select() instead of > >> syncing > >> >> >> with a pacing thread. Which would be dead-quiet at idle time, but, > >> of > >> >> >> course, would stop scaling at some point due to excessive number > of > >> >> >> system calls. > >> >> >> > >> >> >> Thanks. > >> >> >> > >> >> >> > >> >> >> Michael Jerris wrote: > >> >> >>> > >> >> >>> In short. No, you can not for many reasons. The milisecond tic > is > >> >> >>> used throughout the code even when there is not any calls up. > You > >> >> can > >> >> >>> grep for switch_cond_next if you would like to see where but it > is > >> >> >>> required to keep our global timestamp and for pacing the > scheduler > >> >> >>> among other services that run all the time. > >> >> >>> > >> >> >>> Mike > >> >> >>> > >> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >> >>> > >> >> >>>> > >> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >> >> >>>> glanced > >> >> >>>> through the code, and see that among others (are there others?) > >> RTP > >> >> >>>> and IVR > >> >> >>>> set up their timers that are subsequently managed by this > thread. > >> >> >>>> RTP timers > >> >> >>>> should be eliminated by that setting you've suggested. IVR > timers > >> >> >>>> are set at > >> >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead > of > >> >> >>>> 1ms it > >> >> >>>> should be able to wake up those IVR timers just fine. Right? > >> >> >>>> > >> >> >>>> That's a cool design to have one dedicated thread that maintains > >> >> >>>> accurate > >> >> >>>> timing and then broadcasts via condition variables to hundreds > of > >> >> >>>> other > >> >> >>>> threads events that they can register for. I'm sure it's one of > >> the > >> >> >>>> reasons > >> >> >>>> why FS scales so much better than Asterisk. But for poor low-end > >> >> >>>> setups that > >> >> >>>> sit in the closet, eat only 6W of power and hardly ever run more > >> >> >>>> than two > >> >> >>>> calls at the same time, can I hack it somehow to be more UNIX- > >> >> >>>> friendly? I.e. > >> >> >>>> make it stuck in select() or recv() when there is nothing to do, > >> >> call > >> >> >>>> clock_gettime() right from the thread that wants and when it > >> wants > >> >> >>>> to know > >> >> >>>> current time? > >> >> >>>> > >> >> >>>> Say, what if that thread is made to suspend on a condition > >> variable > >> >> >>>> in case > >> >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some > >> >> other > >> >> >>>> thread comes up and adds its timer into the matrix, it could > wake > >> up > >> >> >>>> the > >> >> >>>> timer thread and enjoy accurate timing as needed, on demand? And > >> in- > >> >> >>>> between > >> >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I > >> >> mean, > >> >> >>>> sitting on a wait queue in the kernel is way better than go back > >> and > >> >> >>>> forth > >> >> >>>> incrementing counters that nobody even needs at the moment? > >> >> >>>> > >> >> >>>> > >> >> >>>> Anthony Minessale-2 wrote: > >> >> >>>>> > >> >> >>>>> idle is a 4 letter word to a realtime application. > >> >> >>>>> > >> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing > >> and > >> >> >>>>> expands > >> >> >>>>> that broadcasting > >> >> >>>>> to hundreds or thousand of threads who need accurate timing. > >> >> >>>>> > >> >> >>>>> Your choppy audio is caused by linksys lying about the packet > >> len > >> >> >>>>> that > >> >> >>>>> it's > >> >> >>>>> using and we set our timer > >> >> >>>>> to the wrong speed. > >> >> >>>>> > >> >> >>>>> > >> >> >>>> > >> >> >>>> -- > >> >> >>>> View this message in context: > >> >> >>>> > >> >> > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >> >> >>>> Sent from the Freeswitch-users mailing list archive at > >> Nabble.com. > >> >> >>>> > >> >> >>>> > >> >> >>>> _______________________________________________ > >> >> >>>> FreeSWITCH-users mailing list > >> >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch- > >> >> >>>> users > >> >> >>>> http://www.freeswitch.org > >> >> >>> > >> >> >>> _______________________________________________ > >> >> >>> FreeSWITCH-users mailing list > >> >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >>> UNSUBSCRIBE: > >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org > >> >> >>> > >> >> >>> > >> >> >> > >> >> >> > >> >> > > >> >> > > >> >> > >> >> -- > >> >> View this message in context: > >> >> > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >> > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > -- > >> > Anthony Minessale II > >> > > >> > FreeSWITCH http://www.freeswitch.org/ > >> > ClueCon http://www.cluecon.com/ > >> > Twitter: http://twitter.com/FreeSWITCH_wire > >> > > >> > AIM: anthm > >> > MSN:anthony_minessale at hotmail.com > >> > >< > >> MSN%3Aanthony_minessale at hotmail.com > > > > >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > > > > >> > > >> > IRC: irc.freenode.net #freeswitch > >> > > >> > FreeSWITCH Developer Conference > >> > sip:888 at conference.freeswitch.org > >> > >< > >> sip%3A888 at conference.freeswitch.org > > > > >> > > >> > iax:guest at conference.freeswitch.org/888 > >> > > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> > > > > >> > > >> > pstn:213-799-1400 > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26630994.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b872350e/attachment-0001.html From timuckun at gmail.com Thu Dec 3 12:41:49 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 09:41:49 +1300 Subject: [Freeswitch-users] HA questions. In-Reply-To: <012801ca7439$8cc10ba0$a64322e0$@net> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <012801ca7439$8cc10ba0$a64322e0$@net> Message-ID: <855e4dcf0912031241l73ea409fx71f8e0b7b0b79239@mail.gmail.com> On Fri, Dec 4, 2009 at 5:56 AM, Adam Ford wrote: > Have you checked out Redfone? While I haven't attempted to implement it yet, > my Redfone foneBridge2 claims to be able to handle load balancing and > failover between two Asterisk/Freeswitch servers. > That would be my choice for incoming E1 lines. Right now I am looking for a SIP solution. From dlaperle at rsslex.com Thu Dec 3 13:04:03 2009 From: dlaperle at rsslex.com (David Laperle) Date: Thu, 03 Dec 2009 16:04:03 -0500 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> References: <1259864953.1978.12.camel@dlaplap> <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> Message-ID: <1259874243.8702.9.camel@dlaplap> The files are OK, the permissions on them are OK. Correct me if i'm wrong! If i set the variable "user_context" to "default" it should take into accounts the dialplans into freeswitch/conf/dialplan/default or there's more rules to consider? The Wiki explain that the user must be registered to receive the xml in diaplan/default, my phone is registered (see below) Call-ID: 7cd6e8c8-2c9962a5-7a0da12a at 192.168.102.10 User: 6969@ Contact: "user" Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Status: Registered(UDP)(unknown) EXP(2009-12-03 16:57:35) Host: IP: 192.168.102.10 Port: 5060 Auth-User: unknown Auth-Realm: MWI-Account: 6969@ The only thing i can think of is that my installation of ??FusionPBX?? (installed since day 1) messed something in config files but i really doubt it since i didn't use FusionPBX between yesterday (when was working good) and this morning (after python recompile). The other thing i see is the ??Auth-User?? in my registrations who shows ??unknown?? instead of the actual user ... I'm really lost since everything was working perfectly before my re-compile of mod_python, i think i'll just start all over again ... since i had almost nothing done so far! Thanks for your time, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com On Thu, 2009-12-03 at 14:55 -0500, Michael Collins wrote: > > > > On Thu, Dec 3, 2009 at 10:29 AM, David Laperle > wrote: > > Hi guys, > > i have a weird problem with my dialplans. For the moment, i > have only 2 usable extensions. They were working #1 yesterday, > but this morning i realize i forgot to compile mod_python, so > i go back into my source folder and modify the modules.conf to > uncomment mod_python, did a make and make install (i did a > backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test > dialplan and entered into cli to see what's gonna happen! To > my surprise, the call didn't processed to the extension i was > dialing. > > i tried all the other extensions i had, they were all not > working!!!! > > After that i realized that the .xml in > freeswitch/dialplan/default/ weren't imported into > configuration at startup ... > > I have read all the documentation about difference between > public and default dialplan and i understand them correctly, > in public if i include all default folder, it's working again > (i can reach all my extensions in default. > > My extensions are in the correct user_context ... i did > nothing since yesterday other than a make && make install > after enabling python ... > > Any other user have an idea why the default/*.xml aren't > processed automatically? What could i have done wrong so they > are no longer processed? > > > > double-check for the existence of conf/dialplan/default.xml - I've > seen on rare occasion where that file simple goes away for no apparent > reason. Since I never change that file - and I recommend that you > never change it either ;) - you can go to your FS source directory and > issue "make samples" and it will re-create any missing default config > files without overwriting you existing config files. > -MC > > > -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. Bien que ce courriel ait ?t? trait? contre les virus, il est de la responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot?g?s par le secret professionnel. Si vous d?sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier ?lectronique ordinaire non s?curis?, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c590d9cc/attachment.html From erandr-junk at usa.net Thu Dec 3 13:44:15 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 13:44:15 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> Message-ID: <26633739.post@talk.nabble.com> Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do that. At the moment, I hope it won't be necessary as I can make those "hyper" threads behave, and will see how that goes first. I see where your implementation could be coming from. There is a queue of SQL queries in sofia.c processed by the worker thread. There are only two pop functions available in APR: queue_pop() and queue_trypop(), so alas no option with a timeout here. You don't want to block the thread in pop() indefinitely because you chose that same worker needs to do ireg and gw processing once in a while (separated by tens or hundreds of seconds, btw). You also want to be able to detect shutdown condition so that the worker doesn't hold up profile thread. So you chose to poll for events every millisecond instead of just creating an apr_thread_cond_t for resource friendly signalling. I agree that the timer thread philosophy is great and was the right choice for scaling, but I just don't comprehend responses to things like these other SQL or sofia worker threads. Did somebody even remotely acknowledge that busy loops at least in those areas that I showed may probably be a bad idea and could've been eliminated? I've heard suggestions to bump up priority, I've heard that the code was perfect already, that it's the result of 4-year effort, that I am arrogant, don't listen and don't understand squat. I'm sorry if I gave you impression that I was looking for the bad parts in the software. I apologized for that already. All I wanted was to have constructive conversation, perhaps I'm not too good at it. Code is already perfect according to you? Fine with me. Anthony Minessale-2 wrote: > > no, > > I mean the one after that that you must have completely skipped with a > command line option to try and a param to set in the config. It somewhat > annoys me for taking the time to compose it now. I wrote all of the code > you are talking about myself and I was trying to give you some > suggestions.... > > Well, actually, you did answer my question about the platform so you must > have seen it..... > > The loops are not the cause of that migration message, something wrong > with > the hardware or the kernel is. > Another guy just told you he does not see that problem on the same exact > hardware. > > Even if you have a point about the sql threads, you could make a patch to > slow them down but you cant slow down too much or you will not be able to > handle 400 cps all asking to send updates to transactions in batches of > thousands of sql stmts. Every line of that code is carefully designed so > I > don't know what else to tell you but to stop being so arrogant and re-read > this thread for all the advice you have totally ignored. I started out > trying to help you but I have a lot of work to do. I thoroughly explained > it to you and you are choosing to ignore me so I guess I'm done. > You can do whatever you want with your working copy, i'll see you in 3 or > 4 > years when you get up to speed with the rest of us........ > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mcampbellsmith at gmail.com Thu Dec 3 14:05:35 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 09:05:35 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> Message-ID: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > Check out the Linksys SPA2102 > > On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > wrote: >> >> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> Grandstream HandyTone 503. ?But, again according to the wiki, that >> doesn't seem to behave to well with TLS ... >> >> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >> > Mark Campbell-Smith wrote: >> >> Does the SPA3102 support TLS or only SRTP? >> > >> > I don't know, but supporting only SRTP would be ridiculous, since the >> > keys >> > would then be transmitted in the clear and therefore amenable to >> > interception. >> > SRTP requires the SIP channel to be encrypted by TLS in order to be >> > secure. >> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >> > in >> > RTP. >> > >> > I would be rather surprised were a hardware manufacturer to implement >> > SRTP >> > without TLS for the SIP traffic. On the other hand, we've seen often in >> > this >> > forum that some manufacturers are really clueless... >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From itamar at ispbrasil.com.br Thu Dec 3 14:17:14 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Thu, 3 Dec 2009 20:17:14 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Message-ID: you can try xlite too. On Thu, Dec 3, 2009 at 8:05 PM, Mark Campbell-Smith wrote: > Hi All, > > I managed to borrow a SPA3102 with the latest firmware and have got it > to register using TLS, but I am still struggling with SRTP. ?Has > anyone managed to get SRTP working with the Linksys devices and if so, > can they direct me on how to do this. > > I have generated a mini-certificates and SRTP Private Key using the > gen-mc tool found at > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. > ?However, when ever I initiate a call from the SPA, I can see that the > call is not encrypted. > > Help appreciated. > > Thanks! ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From gkuri at ieee.org Thu Dec 3 14:17:25 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Thu, 3 Dec 2009 14:17:25 -0800 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Message-ID: <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith wrote: > Hi All, > > I managed to borrow a SPA3102 with the latest firmware and have got it > to register using TLS, but I am still struggling with SRTP. ?Has > anyone managed to get SRTP working with the Linksys devices and if so, > can they direct me on how to do this. > > I have generated a mini-certificates and SRTP Private Key using the > gen-mc tool found at > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. > ?However, when ever I initiate a call from the SPA, I can see that the > call is not encrypted. > > Help appreciated. > > Thanks! > > > On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> Check out the Linksys SPA2102 >> >> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> wrote: >>> >>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>> doesn't seem to behave to well with TLS ... >>> >>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>> > Mark Campbell-Smith wrote: >>> >> Does the SPA3102 support TLS or only SRTP? >>> > >>> > I don't know, but supporting only SRTP would be ridiculous, since the >>> > keys >>> > would then be transmitted in the clear and therefore amenable to >>> > interception. >>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>> > secure. >>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>> > in >>> > RTP. >>> > >>> > I would be rather surprised were a hardware manufacturer to implement >>> > SRTP >>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>> > this >>> > forum that some manufacturers are really clueless... >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mcampbellsmith at gmail.com Thu Dec 3 14:34:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 09:34:29 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> Message-ID: <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > exchange to appropriately support SRTP and FreeSWITCH. They do their > proprietary Sipura key exchange only, not sure if Cisco plans on > upgrading the firmware to ever support SDES on the ATAs. They added > support for SDES to their IP Phones about 1 year ago, but nothing has > happened with the ATAs as of yet. > > Gabe > > > On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > wrote: >> Hi All, >> >> I managed to borrow a SPA3102 with the latest firmware and have got it >> to register using TLS, but I am still struggling with SRTP. ?Has >> anyone managed to get SRTP working with the Linksys devices and if so, >> can they direct me on how to do this. >> >> I have generated a mini-certificates and SRTP Private Key using the >> gen-mc tool found at >> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >> ?However, when ever I initiate a call from the SPA, I can see that the >> call is not encrypted. >> >> Help appreciated. >> >> Thanks! >> >> >> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>> Check out the Linksys SPA2102 >>> >>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>> wrote: >>>> >>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>> doesn't seem to behave to well with TLS ... >>>> >>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>> > Mark Campbell-Smith wrote: >>>> >> Does the SPA3102 support TLS or only SRTP? >>>> > >>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>> > keys >>>> > would then be transmitted in the clear and therefore amenable to >>>> > interception. >>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>> > secure. >>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>> > in >>>> > RTP. >>>> > >>>> > I would be rather surprised were a hardware manufacturer to implement >>>> > SRTP >>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>> > this >>>> > forum that some manufacturers are really clueless... >>>> > >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mouncifbb at gmail.com Thu Dec 3 14:33:58 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Thu, 3 Dec 2009 17:33:58 -0500 Subject: [Freeswitch-users] Generate cdrs Message-ID: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/30db44f7/attachment.html From gkuri at ieee.org Thu Dec 3 15:25:29 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Thu, 3 Dec 2009 15:25:29 -0800 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> Message-ID: <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith wrote: > Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > to appropriately support SRTP and FreeSWITCH > > I'll check with Cisco regarding their implementation then and try to > find out when/if they will support standard SRTP encryption. > > > So, back to my origianal question then. ?Are there any ATA's that > support TLS AND SRTP with FreeSwitch? > > > On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> exchange to appropriately support SRTP and FreeSWITCH. They do their >> proprietary Sipura key exchange only, not sure if Cisco plans on >> upgrading the firmware to ever support SDES on the ATAs. They added >> support for SDES to their IP Phones about 1 year ago, but nothing has >> happened with the ATAs as of yet. >> >> Gabe >> >> >> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> wrote: >>> Hi All, >>> >>> I managed to borrow a SPA3102 with the latest firmware and have got it >>> to register using TLS, but I am still struggling with SRTP. ?Has >>> anyone managed to get SRTP working with the Linksys devices and if so, >>> can they direct me on how to do this. >>> >>> I have generated a mini-certificates and SRTP Private Key using the >>> gen-mc tool found at >>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >>> ?However, when ever I initiate a call from the SPA, I can see that the >>> call is not encrypted. >>> >>> Help appreciated. >>> >>> Thanks! >>> >>> >>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>>> Check out the Linksys SPA2102 >>>> >>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>>> wrote: >>>>> >>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>>> doesn't seem to behave to well with TLS ... >>>>> >>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>>> > Mark Campbell-Smith wrote: >>>>> >> Does the SPA3102 support TLS or only SRTP? >>>>> > >>>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>>> > keys >>>>> > would then be transmitted in the clear and therefore amenable to >>>>> > interception. >>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>>> > secure. >>>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>>> > in >>>>> > RTP. >>>>> > >>>>> > I would be rather surprised were a hardware manufacturer to implement >>>>> > SRTP >>>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>>> > this >>>>> > forum that some manufacturers are really clueless... >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From oscav at hotmail.fr Thu Dec 3 15:33:57 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 3 Dec 2009 15:33:57 -0800 (PST) Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: References: <26625147.post@talk.nabble.com> Message-ID: <26635167.post@talk.nabble.com> fs_cli looks like a good idea. I will try that. Many thanks Rob Rob Forman wrote: > > What about cron? > > Create a cron entry like: > */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" > > But if you're just dumping global variables, you could easily retrieve > them > directly from fs_cli without running an app and process the output however > you'd like: > > /usr/local/freeswitch/bin/fs_cli -x "global_getvar" > > > On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > >> >> Hi, >> >> Someone knows how to run periodically a JS script ?? The purpose is to >> write >> to a db some global informations (Global Variables) about FS like every 5 >> minutes. >> >> Thanks. >> >> >> -- >> View this message in context: >> http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26635167.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Dec 3 15:49:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 17:49:20 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26633739.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> Message-ID: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Sigh, You just took it up a notch in terms of disdain and sarcasm. Why do people always only apologize sarcastically? I asked you to try the -hp and turn off the monotonic clock just to gather the results to help you. You completely missed it and just went on about the threads. Please save the "ok fine the code is perfect, blah blah" if you would have just read the email and answered the question I might have cared more about the status of your problem. I told you both of those threads need to be on their toes because they try to balance between a certian number of sql stmts or 500ms whatever comes first. When there are thousands of events per second being turned into SQL statements which are in turn compiled into large sql transactions. If you want to come up with a way that they can sleep longer until there is a sign of activity and stay busy for a few seconds then slow down again, that's probably possible but the process is already idle at 0% cpu so maybe you can appreciate why we are not rushing to work on it. Maybe I'll give it a go just to show you it has nothing to do with your problem. Please don't mock our comment about several years. You have no idea how hard this code was to develop and it's truly insulting. Its clear to see you are locked into assuming that the busy threads that are not all that busy because they are constantly yielding to the scheduler is breaking the timing code. I begged you to understand me when i told you that the err is not normal, most boxes do not see it doing nothing and there has to be a specific problem on your box or configuration. So instead of working with us you want to escalate to snotty comments. That's pretty normal on the internet I guess..... If you want to have a constructive conversation about our core, install FS on a normal box, use it for a few weeks, figure out everything about how it works then try.... There was pure speculation and conjecture in your original emails and I never said a word about it until you kept pushing. Kristian mentioned he never sees that on that same hardware did you even consider following up on why that is? I don't have your device, but I assume if you get it working well it will certainly help you more than it helps me so you could at least have the decency to believe what we are trying to tell you. On Thu, Dec 3, 2009 at 3:44 PM, eaf wrote: > > Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do > that. > At the moment, I hope it won't be necessary as I can make those "hyper" > threads behave, and will see how that goes first. I see where your > implementation could be coming from. There is a queue of SQL queries in > sofia.c processed by the worker thread. There are only two pop functions > available in APR: queue_pop() and queue_trypop(), so alas no option with a > timeout here. You don't want to block the thread in pop() indefinitely > because you chose that same worker needs to do ireg and gw processing once > in a while (separated by tens or hundreds of seconds, btw). You also want > to > be able to detect shutdown condition so that the worker doesn't hold up > profile thread. So you chose to poll for events every millisecond instead > of > just creating an apr_thread_cond_t for resource friendly signalling. > > I agree that the timer thread philosophy is great and was the right choice > for scaling, but I just don't comprehend responses to things like these > other SQL or sofia worker threads. Did somebody even remotely acknowledge > that busy loops at least in those areas that I showed may probably be a bad > idea and could've been eliminated? I've heard suggestions to bump up > priority, I've heard that the code was perfect already, that it's the > result > of 4-year effort, that I am arrogant, don't listen and don't understand > squat. > > I'm sorry if I gave you impression that I was looking for the bad parts in > the software. I apologized for that already. All I wanted was to have > constructive conversation, perhaps I'm not too good at it. Code is already > perfect according to you? Fine with me. > > > Anthony Minessale-2 wrote: > > > > no, > > > > I mean the one after that that you must have completely skipped with a > > command line option to try and a param to set in the config. It somewhat > > annoys me for taking the time to compose it now. I wrote all of the code > > you are talking about myself and I was trying to give you some > > suggestions.... > > > > Well, actually, you did answer my question about the platform so you > must > > have seen it..... > > > > The loops are not the cause of that migration message, something wrong > > with > > the hardware or the kernel is. > > Another guy just told you he does not see that problem on the same exact > > hardware. > > > > Even if you have a point about the sql threads, you could make a patch to > > slow them down but you cant slow down too much or you will not be able to > > handle 400 cps all asking to send updates to transactions in batches of > > thousands of sql stmts. Every line of that code is carefully designed so > > I > > don't know what else to tell you but to stop being so arrogant and > re-read > > this thread for all the advice you have totally ignored. I started out > > trying to help you but I have a lot of work to do. I thoroughly > explained > > it to you and you are choosing to ignore me so I guess I'm done. > > You can do whatever you want with your working copy, i'll see you in 3 or > > 4 > > years when you get up to speed with the rest of us........ > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/538b8cbc/attachment-0001.html From dujinfang at gmail.com Thu Dec 3 16:02:53 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 08:02:53 +0800 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: References: Message-ID: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane : > is it possible to run a javascript at the end of dialplan to generate cdrs? > because (mod_cdr_csv) is giving me hard time as it rotates Master file on > machine reboots or shutdown signals. > javascript or LUA for preferences? > > thank you > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Thu Dec 3 16:06:29 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 08:06:29 +0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B180525.7060702@greatiam.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> <4B180525.7060702@greatiam.com> Message-ID: <23f91030912031606m4cc2698dyb53a2b22754d05ef@mail.gmail.com> You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ? 2009/12/4 Samuel Abekah-Mensah : > Hi > > Sorry .xm is a typo. I actually shut down the server and restarted. The > log says I need to create a domain of aaa.bbb.ccc.ddd (which is the > server IP address ) and then put the user in that domain. ?Isn't the > default domain that of the server FS is running on ? > 2319.xml is in /usr/local/freeswitch/conf/directory/default/ > > Thanks for your time > > > > Michael Collins wrote: >> >> >> On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah >> > wrote: >> >> ? ? I have copied 1001.xml in directory/default to a test user 2319.xm >> ? ? changing or instances of 1001 in the file to 2319. I then went into >> ? ? default.xml ?in directory folder and in one of the groups ?just >> ? ? mimicked >> ? ? 1001 details by changing 1001 to 2319. >> >> ? ? Connecting ?to FS gives Forbidden message. However 1001 connects >> ? ? without >> ? ? a problem. ?What have I missed ? >> >> ? ? Is there a place that just puts things in do this and that and that to >> ? ? create a new user ? >> >> >> Did you execute "reloadxml" from the fs cli before trying to connect >> with 2319? Also I'm assuming that "2319.xm" is a typo and you actually >> created "2319.xml" in the default/directory subdir. >> -MC >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From timuckun at gmail.com Thu Dec 3 16:22:53 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 13:22:53 +1300 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <26625105.post@talk.nabble.com> References: <26625105.post@talk.nabble.com> Message-ID: <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> > > Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) > ports fail being opened dynamically to work properly, or does SIP today > really work well over NAT firewalls? > Yes I get issues quite a bit with the server being behind a firewall. IAX is much nicer in this circumstance. From jason at jasonjgw.net Thu Dec 3 16:35:24 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 4 Dec 2009 11:35:24 +1100 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> References: <26625105.post@talk.nabble.com> <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> Message-ID: <20091204003524.GA22701@jdc.jasonjgw.net> Tim Uckun wrote: > Yes I get issues quite a bit with the server being behind a firewall. > IAX is much nicer in this circumstance. I just set up an IPv6 over IPv4 tunnel and nat goes away. I have native IPv6 over ADSL now, as part of a trial that my ISP is conducting. As a result, one end of the conection doesn't go through a tunnel provider anymore. Given the problems I've had (and still have) with nat, I want to be rid of it as much as possible. Nevertheless, I agree that in a nat scenario, IAX can be easier to configure correctly. From mike at jerris.com Thu Dec 3 16:56:24 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 19:56:24 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> Message-ID: <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover. Mike On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: > On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: >> The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? >> > > From an as of now unkown SIP trunk provider (we are still in > negotiations with a couple of companies). > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Dec 3 17:47:35 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Dec 2009 20:47:35 -0500 Subject: [Freeswitch-users] Playing an rtp stream Message-ID: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> Hi there, It it possible do something like: Basically I have need to connect to incoming calls listen to an existing rtp stream - I know the IP and port. Any hints on achieving this would be much appreciated. Thanks Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/71b9ea52/attachment.html From mcampbellsmith at gmail.com Thu Dec 3 18:26:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 13:26:17 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> Message-ID: <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > Grandstream and Mediatrix devices (although I've never tried either > one with FreeSWITCH). > > I've personally never had any good experience with the Grandstream > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > support it (so says their marketing material and docs). > > I'd see if Cisco has any plans to add support for it to the ATAs. Next > time I see our Cisco SE, I'll try to poke him about it. > > Gabe > > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > wrote: >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange >> to appropriately support SRTP and FreeSWITCH >> >> I'll check with Cisco regarding their implementation then and try to >> find out when/if they will support standard SRTP encryption. >> >> >> So, back to my origianal question then. ?Are there any ATA's that >> support TLS AND SRTP with FreeSwitch? >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >>> exchange to appropriately support SRTP and FreeSWITCH. They do their >>> proprietary Sipura key exchange only, not sure if Cisco plans on >>> upgrading the firmware to ever support SDES on the ATAs. They added >>> support for SDES to their IP Phones about 1 year ago, but nothing has >>> happened with the ATAs as of yet. >>> >>> Gabe >>> >>> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >>> wrote: >>>> Hi All, >>>> >>>> I managed to borrow a SPA3102 with the latest firmware and have got it >>>> to register using TLS, but I am still struggling with SRTP. ?Has >>>> anyone managed to get SRTP working with the Linksys devices and if so, >>>> can they direct me on how to do this. >>>> >>>> I have generated a mini-certificates and SRTP Private Key using the >>>> gen-mc tool found at >>>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >>>> ?However, when ever I initiate a call from the SPA, I can see that the >>>> call is not encrypted. >>>> >>>> Help appreciated. >>>> >>>> Thanks! >>>> >>>> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>>>> Check out the Linksys SPA2102 >>>>> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>>>> wrote: >>>>>> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>>>> doesn't seem to behave to well with TLS ... >>>>>> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>>>> > Mark Campbell-Smith wrote: >>>>>> >> Does the SPA3102 support TLS or only SRTP? >>>>>> > >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>>>> > keys >>>>>> > would then be transmitted in the clear and therefore amenable to >>>>>> > interception. >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>>>> > secure. >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>>>> > in >>>>>> > RTP. >>>>>> > >>>>>> > I would be rather surprised were a hardware manufacturer to implement >>>>>> > SRTP >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>>>> > this >>>>>> > forum that some manufacturers are really clueless... >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Thu Dec 3 20:38:21 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 06:38:21 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Message-ID: Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith > Cheers Gabriel.. thanks for the information. > > I'll look at the Mediatrix ATA's as an alternative - has anyone had > experience with those and TLS/SRTP? > > > On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > > Grandstream and Mediatrix devices (although I've never tried either > > one with FreeSWITCH). > > > > I've personally never had any good experience with the Grandstream > > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > > support it (so says their marketing material and docs). > > > > I'd see if Cisco has any plans to add support for it to the ATAs. Next > > time I see our Cisco SE, I'll try to poke him about it. > > > > Gabe > > > > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > > wrote: > >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > >> to appropriately support SRTP and FreeSWITCH > >> > >> I'll check with Cisco regarding their implementation then and try to > >> find out when/if they will support standard SRTP encryption. > >> > >> > >> So, back to my origianal question then. Are there any ATA's that > >> support TLS AND SRTP with FreeSwitch? > >> > >> > >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > >>> exchange to appropriately support SRTP and FreeSWITCH. They do their > >>> proprietary Sipura key exchange only, not sure if Cisco plans on > >>> upgrading the firmware to ever support SDES on the ATAs. They added > >>> support for SDES to their IP Phones about 1 year ago, but nothing has > >>> happened with the ATAs as of yet. > >>> > >>> Gabe > >>> > >>> > >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > >>> wrote: > >>>> Hi All, > >>>> > >>>> I managed to borrow a SPA3102 with the latest firmware and have got it > >>>> to register using TLS, but I am still struggling with SRTP. Has > >>>> anyone managed to get SRTP working with the Linksys devices and if so, > >>>> can they direct me on how to do this. > >>>> > >>>> I have generated a mini-certificates and SRTP Private Key using the > >>>> gen-mc tool found at > >>>> > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 > . > >>>> However, when ever I initiate a call from the SPA, I can see that the > >>>> call is not encrypted. > >>>> > >>>> Help appreciated. > >>>> > >>>> Thanks! > >>>> > >>>> > >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > >>>>> Check out the Linksys SPA2102 > >>>>> > >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > >>>>> wrote: > >>>>>> > >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the > >>>>>> Grandstream HandyTone 503. But, again according to the wiki, that > >>>>>> doesn't seem to behave to well with TLS ... > >>>>>> > >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White > wrote: > >>>>>> > Mark Campbell-Smith wrote: > >>>>>> >> Does the SPA3102 support TLS or only SRTP? > >>>>>> > > >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since > the > >>>>>> > keys > >>>>>> > would then be transmitted in the clear and therefore amenable to > >>>>>> > interception. > >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to > be > >>>>>> > secure. > >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works > entirely > >>>>>> > in > >>>>>> > RTP. > >>>>>> > > >>>>>> > I would be rather surprised were a hardware manufacturer to > implement > >>>>>> > SRTP > >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen > often in > >>>>>> > this > >>>>>> > forum that some manufacturers are really clueless... > >>>>>> > > >>>>>> > > >>>>>> > _______________________________________________ > >>>>>> > FreeSWITCH-users mailing list > >>>>>> > FreeSWITCH-users at lists.freeswitch.org > >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> > http://www.freeswitch.org > >>>>>> > > >>>>>> > >>>>>> _______________________________________________ > >>>>>> FreeSWITCH-users mailing list > >>>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/6ed9e6a3/attachment-0001.html From yehavi.bourvine at gmail.com Thu Dec 3 20:40:25 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 06:40:25 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: References: <4B1800C7.7010800@metik.com> Message-ID: I am taking my words back... The Cisco sends back what I want. I got confused because the Nortel sends the name only for the connected PBX and not for the othes ones (although it gets this infomation from them). Thanks, __Yehavi: 2009/12/3 Yehavi Bourvine > Unfortunately this didn't help... Incoming calls from ISDN to SIP sends > back to ISDN the name of the destination, but not the other way around... > > Thanks! __Yehavi: > > 2009/12/3 Metik > > Yehavi, >> >> There are a few variations of transmitting this information... If you >> have already enabled a supplemental isdn service profile, try adding the >> following to the PRI you are using: >> >> (config-if)#isdn outgoing ie facility >> (config-if)#iisdn outgoing ie extended-facility >> (config-if)#isdn outgoing display-ie >> (config-if)#isdn outgoing ie caller-number >> (config-if)#isdn outgoing ie called-number >> >> -metik >> >> Yehavi Bourvine wrote: >> > Hello, >> > >> > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On >> > the PRI there is a Nortel with Q.Sig. After a lot of configuration >> > trials I've managed to set it to send back the connected name over the >> > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the >> > connected name and then the Cisco adds it as a Remote-Party-ID). >> > However, I did not save it and a power outage cleared this config. In >> > my age I don't remember what I've done... >> > >> > Anyone knows the correct config? >> > >> > Thanks! __Yehavi: >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/70f2cadc/attachment.html From mcampbellsmith at gmail.com Thu Dec 3 21:01:21 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 16:01:21 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Message-ID: <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks! On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine wrote: > Hello, > > ? I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they > should support TLS also (will try it next week; up to now I preffered to not > use TLS so I can sniff the traffic and debug things). > > ???????????????? Regards, __Yehavi: > > 2009/12/4 Mark Campbell-Smith >> >> Cheers Gabriel.. thanks for the information. >> >> I'll look at the Mediatrix ATA's as an alternative - has anyone had >> experience with those and TLS/SRTP? >> >> >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the >> > Grandstream and Mediatrix devices (although I've never tried either >> > one with FreeSWITCH). >> > >> > I've personally never had any good experience with the Grandstream >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they >> > support it (so says their marketing material and docs). >> > >> > I'd see if Cisco has any plans to add support for it to the ATAs. Next >> > time I see our Cisco SE, I'll try to poke him about it. >> > >> > Gabe >> > >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith >> > wrote: >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange >> >> to appropriately support SRTP and FreeSWITCH >> >> >> >> I'll check with Cisco regarding their implementation then and try to >> >> find out when/if they will support standard SRTP encryption. >> >> >> >> >> >> So, back to my origianal question then. ?Are there any ATA's that >> >> support TLS AND SRTP with FreeSwitch? >> >> >> >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do their >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on >> >>> upgrading the firmware to ever support SDES on the ATAs. They added >> >>> support for SDES to their IP Phones about 1 year ago, but nothing has >> >>> happened with the ATAs as of yet. >> >>> >> >>> Gabe >> >>> >> >>> >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> >>> wrote: >> >>>> Hi All, >> >>>> >> >>>> I managed to borrow a SPA3102 with the latest firmware and have got >> >>>> it >> >>>> to register using TLS, but I am still struggling with SRTP. ?Has >> >>>> anyone managed to get SRTP working with the Linksys devices and if >> >>>> so, >> >>>> can they direct me on how to do this. >> >>>> >> >>>> I have generated a mini-certificates and SRTP Private Key using the >> >>>> gen-mc tool found at >> >>>> >> >>>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >> >>>> ?However, when ever I initiate a call from the SPA, I can see that >> >>>> the >> >>>> call is not encrypted. >> >>>> >> >>>> Help appreciated. >> >>>> >> >>>> Thanks! >> >>>> >> >>>> >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> >>>>> Check out the Linksys SPA2102 >> >>>>> >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> >>>>> wrote: >> >>>>>> >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> >>>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >> >>>>>> doesn't seem to behave to well with TLS ... >> >>>>>> >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White >> >>>>>> wrote: >> >>>>>> > Mark Campbell-Smith wrote: >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? >> >>>>>> > >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since >> >>>>>> > the >> >>>>>> > keys >> >>>>>> > would then be transmitted in the clear and therefore amenable to >> >>>>>> > interception. >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to >> >>>>>> > be >> >>>>>> > secure. >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works >> >>>>>> > entirely >> >>>>>> > in >> >>>>>> > RTP. >> >>>>>> > >> >>>>>> > I would be rather surprised were a hardware manufacturer to >> >>>>>> > implement >> >>>>>> > SRTP >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen >> >>>>>> > often in >> >>>>>> > this >> >>>>>> > forum that some manufacturers are really clueless... >> >>>>>> > >> >>>>>> > >> >>>>>> > _______________________________________________ >> >>>>>> > FreeSWITCH-users mailing list >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>> > >> >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>> > http://www.freeswitch.org >> >>>>>> > >> >>>>>> >> >>>>>> _______________________________________________ >> >>>>>> FreeSWITCH-users mailing list >> >>>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>> >> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>> http://www.freeswitch.org >> >>>>> >> >>>>> >> >>>>> _______________________________________________ >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>> http://www.freeswitch.org >> >>>>> >> >>>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From neilp at cs.stanford.edu Thu Dec 3 22:28:50 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 11:58:50 +0530 Subject: [Freeswitch-users] record mp3s Message-ID: Hi All, This is a great list, thanks for all of the support! For my IVR app running on FS, we we accept potentially long audio recordings. Is it possible (in lua) to save recorded as mp3? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/0d1233ec/attachment.html From mrene_lists at avgs.ca Thu Dec 3 22:31:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 01:31:12 -0500 Subject: [Freeswitch-users] record mp3s In-Reply-To: References: Message-ID: <8FADAED2-E609-4F01-B81F-010E242A9F0A@avgs.ca> Hi Neil, If you have mod_shout loaded and use a .mp3 file as you recording filename, it'll automagically encode it. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 1:28 AM, Neil Patel wrote: > Hi All, > > This is a great list, thanks for all of the support! > > For my IVR app running on FS, we we accept potentially long audio > recordings. Is it possible (in lua) to save recorded as mp3? > > Thanks, > Neil > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From neilp at cs.stanford.edu Thu Dec 3 22:36:04 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 12:06:04 +0530 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: Thanks all for your help. I got around this by running ./Setup and installing wanpipe in TDM API mode (it says it's the default for FS). I then uncommented the mod_openzap line in modules.conf when installing FS. Finally I ran wancfg_fs which creates appropriate config files for you for your FS installation. I believe openzap is now installed properly: 2009-12-04 12:04:52.411017 [INFO] zap_io.c:2451 Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2009-12-04 12:04:52.411126 [INFO] zap_io.c:2251 auto-loaded 'wanpipe' 2009-12-04 12:04:52.411311 [INFO] ozmod_wanpipe.c:287 configuring device s1c1 as OpenZAP device 1:1 fd:14 DTMF: software 2009-12-04 12:04:52.411377 [INFO] ozmod_wanpipe.c:287 configuring device s1c2 as OpenZAP device 1:2 fd:15 DTMF: software 2009-12-04 12:04:52.411444 [INFO] ozmod_wanpipe.c:287 configuring device s1c3 as OpenZAP device 1:3 fd:17 DTMF: software 2009-12-04 12:04:52.411509 [INFO] ozmod_wanpipe.c:287 configuring device s1c4 as OpenZAP device 1:4 fd:18 DTMF: software 2009-12-04 12:04:52.411575 [INFO] ozmod_wanpipe.c:287 configuring device s1c5 as OpenZAP device 1:5 fd:19 DTMF: software 2009-12-04 12:04:52.411639 [INFO] ozmod_wanpipe.c:287 configuring device s1c6 as OpenZAP device 1:6 fd:20 DTMF: software 2009-12-04 12:04:52.411707 [INFO] ozmod_wanpipe.c:287 configuring device s1c7 as OpenZAP device 1:7 fd:21 DTMF: software 2009-12-04 12:04:52.411771 [INFO] ozmod_wanpipe.c:287 configuring device s1c8 as OpenZAP device 1:8 fd:22 DTMF: software 2009-12-04 12:04:52.411837 [INFO] ozmod_wanpipe.c:287 configuring device s1c9 as OpenZAP device 1:9 fd:23 DTMF: software 2009-12-04 12:04:52.411903 [INFO] ozmod_wanpipe.c:287 configuring device s1c10 as OpenZAP device 1:10 fd:24 DTMF: software 2009-12-04 12:04:52.411969 [INFO] ozmod_wanpipe.c:287 configuring device s1c11 as OpenZAP device 1:11 fd:25 DTMF: software 2009-12-04 12:04:52.412034 [INFO] ozmod_wanpipe.c:287 configuring device s1c12 as OpenZAP device 1:12 fd:26 DTMF: software 2009-12-04 12:04:52.412102 [INFO] ozmod_wanpipe.c:287 configuring device s1c13 as OpenZAP device 1:13 fd:27 DTMF: software 2009-12-04 12:04:52.412179 [INFO] ozmod_wanpipe.c:287 configuring device s1c14 as OpenZAP device 1:14 fd:28 DTMF: software 2009-12-04 12:04:52.412244 [INFO] ozmod_wanpipe.c:287 configuring device s1c15 as OpenZAP device 1:15 fd:29 DTMF: software TDM API: CMD: 18 : Operation not supported 2009-12-04 12:04:52.412416 [INFO] ozmod_wanpipe.c:287 configuring device s1c16 as OpenZAP device 1:16 fd:30 DTMF: none 2009-12-04 12:04:52.412503 [INFO] ozmod_wanpipe.c:287 configuring device s1c17 as OpenZAP device 1:17 fd:31 DTMF: software 2009-12-04 12:04:52.412568 [INFO] ozmod_wanpipe.c:287 configuring device s1c18 as OpenZAP device 1:18 fd:32 DTMF: software 2009-12-04 12:04:52.412634 [INFO] ozmod_wanpipe.c:287 configuring device s1c19 as OpenZAP device 1:19 fd:33 DTMF: software 2009-12-04 12:04:52.412708 [INFO] ozmod_wanpipe.c:287 configuring device s1c20 as OpenZAP device 1:20 fd:34 DTMF: software 2009-12-04 12:04:52.412771 [INFO] ozmod_wanpipe.c:287 configuring device s1c21 as OpenZAP device 1:21 fd:35 DTMF: software 2009-12-04 12:04:52.412838 [INFO] ozmod_wanpipe.c:287 configuring device s1c22 as OpenZAP device 1:22 fd:36 DTMF: software 2009-12-04 12:04:52.412902 [INFO] ozmod_wanpipe.c:287 configuring device s1c23 as OpenZAP device 1:23 fd:37 DTMF: software 2009-12-04 12:04:52.412948 [INFO] ozmod_wanpipe.c:287 configuring device s1c24 as OpenZAP device 1:24 fd:38 DTMF: software 2009-12-04 12:04:52.412988 [INFO] ozmod_wanpipe.c:287 configuring device s1c25 as OpenZAP device 1:25 fd:39 DTMF: software 2009-12-04 12:04:52.413018 [INFO] ozmod_wanpipe.c:287 configuring device s1c26 as OpenZAP device 1:26 fd:40 DTMF: software 2009-12-04 12:04:52.413041 [INFO] ozmod_wanpipe.c:287 configuring device s1c27 as OpenZAP device 1:27 fd:41 DTMF: software 2009-12-04 12:04:52.413063 [INFO] ozmod_wanpipe.c:287 configuring device s1c28 as OpenZAP device 1:28 fd:42 DTMF: software 2009-12-04 12:04:52.413086 [INFO] ozmod_wanpipe.c:287 configuring device s1c29 as OpenZAP device 1:29 fd:43 DTMF: software 2009-12-04 12:04:52.413106 [INFO] ozmod_wanpipe.c:287 configuring device s1c30 as OpenZAP device 1:30 fd:44 DTMF: software 2009-12-04 12:04:52.413128 [INFO] ozmod_wanpipe.c:287 configuring device s1c31 as OpenZAP device 1:31 fd:45 DTMF: software 2009-12-04 12:04:52.413142 [INFO] zap_io.c:2374 Configured 31 channel(s) 2009-12-04 12:04:52.431405 [INFO] zap_io.c:2468 Loading SIG from /usr/local/freeswitch/mod/ozmod_ss7_boost.so 2009-12-04 12:04:52.431441 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost' 2009-12-04 12:04:52.431541 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_openzap] 2009-12-04 12:04:52.431553 [NOTICE] switch_loadable_module.c:142 Adding Endpoint 'openzap' 2009-12-04 12:04:52.431638 [NOTICE] switch_loadable_module.c:248 Adding Application 'disable_ec' 2009-12-04 12:04:52.431659 [NOTICE] switch_loadable_module.c:270 Adding API Function 'oz' 2009-12-04 12:04:52.432009 [WARNING] ss7_boost_client.c:244 TX EVENT (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0 Does this look right? Thanks. On Mon, Nov 30, 2009 at 9:09 PM, Moises Silva wrote: > On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: > >> Hi All, >> >> I am currently installing a Sangoma A102 card to work with FS using >> wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get >> openzap-related modules to compile: >> >> > cd wanpipe-3.5.6.5/ >> > make openzap >> ... >> make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >> make -C api/libstelephony clean >> make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >> make[1]: *** No rule to make target `clean'. Stop. >> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >> make: *** [all_lib] Error 2 >> >> The libstelephony directory has no Makefile in it. Why is it missing? Is >> there a version of wanpipe drivers that will work? I have been unsuccessful >> with 3.4.4 and 3.5.6 in similar fashion. >> >> > Hi Neil, > > Most likely the creation of the Makefile failed (since you mention you > can't see a Makefile). Please be sure to have installed the pre-requisites > listed at http://wiki.sangoma.com/Requirements > > Particularly in this case, libtool, autoconf and automake packages. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bf531d7a/attachment-0001.html From jbr at consiglia.dk Thu Dec 3 22:40:22 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 4 Dec 2009 07:40:22 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: Anthony, you advised me to use MySQL as the core database in order to access it from Lua. I'm testing that as a work-around. Still, I guess that your choice of SQLite as the default core database have been taken from efficiency or stability considerations. Using MySQL through an ODBC-connector does not sound as a clean solution. Have you any experience about "how bad" it is to use the ODBC MySQL combination in terms of stability, memory leaks and efficiency? Regards Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/333d0202/attachment.html From mrene_lists at avgs.ca Thu Dec 3 22:42:34 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 01:42:34 -0500 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <3E4F23C1-9835-497C-B587-645E4B53F043@avgs.ca> ODBC isnt as bad as its used to be. We use it with postgresql every day and are very happy with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 1:40 AM, Jon Bruel wrote: > Anthony, you advised me to use MySQL as the core database in order > to access it from Lua. I?m testing that as a work-around. > > Still, I guess that your choice of SQLite as the default core > database have been taken from efficiency or stability > considerations. Using MySQL through an ODBC-connector does not sound > as a clean solution. Have you any experience about ?how bad? it is > to use the ODBC MySQL combination in terms of stability, memory > leaks and efficiency? > > Regards > > Jon Br?el > Consiglia Telecommunications > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > CVR: 27047882 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/60d2305a/attachment.html From yehavi.bourvine at gmail.com Fri Dec 4 01:19:51 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 11:19:51 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: I'll report when I am done. So far I've enabled only SRTP and both support it. __Yehavi: 2009/12/4 Mark Campbell-Smith > Thanks Yehavi, > > I would be very interested to find out how your test goes... can you > report back after you have tested it? > > Thanks! > > On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine > wrote: > > Hello, > > > > I have AudioCodes MP and Vega ATA adapters. They both support SRTP; > they > > should support TLS also (will try it next week; up to now I preffered to > not > > use TLS so I can sniff the traffic and debug things). > > > > Regards, __Yehavi: > > > > 2009/12/4 Mark Campbell-Smith > >> > >> Cheers Gabriel.. thanks for the information. > >> > >> I'll look at the Mediatrix ATA's as an alternative - has anyone had > >> experience with those and TLS/SRTP? > >> > >> > >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > >> > Grandstream and Mediatrix devices (although I've never tried either > >> > one with FreeSWITCH). > >> > > >> > I've personally never had any good experience with the Grandstream > >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > >> > support it (so says their marketing material and docs). > >> > > >> > I'd see if Cisco has any plans to add support for it to the ATAs. Next > >> > time I see our Cisco SE, I'll try to poke him about it. > >> > > >> > Gabe > >> > > >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > >> > wrote: > >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > >> >> to appropriately support SRTP and FreeSWITCH > >> >> > >> >> I'll check with Cisco regarding their implementation then and try to > >> >> find out when/if they will support standard SRTP encryption. > >> >> > >> >> > >> >> So, back to my origianal question then. Are there any ATA's that > >> >> support TLS AND SRTP with FreeSwitch? > >> >> > >> >> > >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do their > >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on > >> >>> upgrading the firmware to ever support SDES on the ATAs. They added > >> >>> support for SDES to their IP Phones about 1 year ago, but nothing > has > >> >>> happened with the ATAs as of yet. > >> >>> > >> >>> Gabe > >> >>> > >> >>> > >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > >> >>> wrote: > >> >>>> Hi All, > >> >>>> > >> >>>> I managed to borrow a SPA3102 with the latest firmware and have got > >> >>>> it > >> >>>> to register using TLS, but I am still struggling with SRTP. Has > >> >>>> anyone managed to get SRTP working with the Linksys devices and if > >> >>>> so, > >> >>>> can they direct me on how to do this. > >> >>>> > >> >>>> I have generated a mini-certificates and SRTP Private Key using the > >> >>>> gen-mc tool found at > >> >>>> > >> >>>> > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 > . > >> >>>> However, when ever I initiate a call from the SPA, I can see that > >> >>>> the > >> >>>> call is not encrypted. > >> >>>> > >> >>>> Help appreciated. > >> >>>> > >> >>>> Thanks! > >> >>>> > >> >>>> > >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > >> >>>>> Check out the Linksys SPA2102 > >> >>>>> > >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > >> >>>>> wrote: > >> >>>>>> > >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the > >> >>>>>> Grandstream HandyTone 503. But, again according to the wiki, > that > >> >>>>>> doesn't seem to behave to well with TLS ... > >> >>>>>> > >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White > > >> >>>>>> wrote: > >> >>>>>> > Mark Campbell-Smith wrote: > >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? > >> >>>>>> > > >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, > since > >> >>>>>> > the > >> >>>>>> > keys > >> >>>>>> > would then be transmitted in the clear and therefore amenable > to > >> >>>>>> > interception. > >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order > to > >> >>>>>> > be > >> >>>>>> > secure. > >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works > >> >>>>>> > entirely > >> >>>>>> > in > >> >>>>>> > RTP. > >> >>>>>> > > >> >>>>>> > I would be rather surprised were a hardware manufacturer to > >> >>>>>> > implement > >> >>>>>> > SRTP > >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen > >> >>>>>> > often in > >> >>>>>> > this > >> >>>>>> > forum that some manufacturers are really clueless... > >> >>>>>> > > >> >>>>>> > > >> >>>>>> > _______________________________________________ > >> >>>>>> > FreeSWITCH-users mailing list > >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org > >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>> > > >> >>>>>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>> > http://www.freeswitch.org > >> >>>>>> > > >> >>>>>> > >> >>>>>> _______________________________________________ > >> >>>>>> FreeSWITCH-users mailing list > >> >>>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>> > >> >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>> http://www.freeswitch.org > >> >>>>> > >> >>>>> > >> >>>>> _______________________________________________ > >> >>>>> FreeSWITCH-users mailing list > >> >>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>> > >> >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>> http://www.freeswitch.org > >> >>>>> > >> >>>>> > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>> http://www.freeswitch.org > >> >>>> > >> >>> > >> >>> _______________________________________________ > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bc9e1245/attachment-0001.html From jbr at consiglia.dk Fri Dec 4 01:24:01 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 4 Dec 2009 10:24:01 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I have now tested the FS with core db configured using MySql (by modifying the switch.conf.xml file). Unfortunately, it does not solve my problem because some of the core tables still remain as active SQLite tables. After restarting the FS in the new configuration (with SQLite database core deleted), the following tables are created in MySql and SQLite: MySQL: aliases, complete, nat and tasks (database starting with no tables prior to FS restart). SQLite: aliases, calls, channels, interfaces, nat and tasks. As I would like to access the channels table using Lua, the change did not fix my problem. I have positive verified that the channels table is active and populated during calls. Are there other places where I should define the usage of the MySql database? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/99e4251a/attachment.html From codecomplete at free.fr Fri Dec 4 02:32:14 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 4 Dec 2009 02:32:14 -0800 (PST) Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> References: <26625105.post@talk.nabble.com> <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> Message-ID: <26635842.post@talk.nabble.com> Michael Jerris wrote: > with a client that does not support stun or at least rfc 3581 the results > are much more sketchy and require more hacks on the server side, but with > enough effort can almost always be made to work. Thanks Mike for the feedback. If a user has a problem using my FS server, I'll check what client they have. For those interested, here's what RFC 3581 adds to SIP: "Session Initiation Protocol (SIP) operates over UDP and TCP, among others. When used with UDP, responses to requests are returned to the source address the request came from, and to the port written into the topmost Via header field value of the request. This behavior is not desirable in many cases, most notably, when the client is behind a Network Address Translator (NAT). This extension defines a new parameter for the Via header field, called "rport", that allows a client to request that the server send the response back to the source IP address and port from which the request originated." -- View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26635842.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Fri Dec 4 04:39:10 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 4 Dec 2009 13:39:10 +0100 Subject: [Freeswitch-users] HA questions. In-Reply-To: <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> Message-ID: <65d96fc80912040439x727f2a0jdb6bd07a41971f0@mail.gmail.com> Hi Mike, Lets suppose we have: - 2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. ( http://www.ultramonkey.org/3/topologies/ha-overview.html) - freeswitch installed on every machine configured to use mysql in the core via odbc - both freeswitch have identical dialplan and directory configuration - mysql installed on every machine (with replication between the DBs) - SIP Trunks towards the upper provider (without registration but i should work with registration) - SIP Phones/Terminals registering to the active freeswitch When a terminal registers to the active freeswitch, the registration is propagated to the inactive one via DB replication. Now, lets suppose we have a switchover ... of course we will lose the ongoing calls but new calls (from SIP Phones) should be able to establish. The same applies to incoming calls from the upper provider. Im just talking about HA here not loadbalancing and performance scaling... what do you think about that? On Fri, Dec 4, 2009 at 1:56 AM, Michael Jerris wrote: > so your registering to the provider to get the calls? If so, this gets > tricky, the provider likely does not support multiple registrations, even if > they did they probably send the call to both registered endpoints. With > this big unknown its not very easy to suggest a good solution. If I were > looking to set this up without needing proxies I would want to use srv > records and naptr records and a provider that would balance using these > including failiover. > > Mike > > > On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: > > > On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: > >> The easiest place to do this is at the point you send the calls to > FreeSWITCH. How are the calls coming in? > >> > > > > From an as of now unkown SIP trunk provider (we are still in > > negotiations with a couple of companies). > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/ca6d2e10/attachment.html From mouncifbb at gmail.com Fri Dec 4 06:33:53 2009 From: mouncifbb at gmail.com (Mouncifbb) Date: Fri, 4 Dec 2009 09:33:53 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> Message-ID: <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> I don't want to use XML cdr it puts each call on individual files so is it possible to include a JavaScript at the end of dialplan to collect info about the session? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > why not try mod_xml_cdr? > > 2009/12/4 Mouncif Benniane : >> is it possible to run a javascript at the end of dialplan to >> generate cdrs? >> because (mod_cdr_csv) is giving me hard time as it rotates Master >> file on >> machine reboots or shutdown signals. >> javascript or LUA for preferences? >> >> thank you >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Fri Dec 4 06:59:27 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 22:59:27 +0800 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> Message-ID: <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> 2009/12/4 Mouncifbb : > I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails > is it possible to include a JavaScript at the end of dialplan to > collect info about the session? > I think the answer is yes but where would you store the collected info? > Thanks > > > On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > >> why not try mod_xml_cdr? >> >> 2009/12/4 Mouncif Benniane : >>> is it possible to run a javascript at the end of dialplan to >>> generate cdrs? >>> because (mod_cdr_csv) is giving me hard time as it rotates Master >>> file on >>> machine reboots or shutdown signals. >>> javascript or LUA for preferences? >>> >>> thank you >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mouncifbb at gmail.com Fri Dec 4 07:14:24 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Fri, 4 Dec 2009 10:14:24 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> Message-ID: I wanna store it on different file out of cdr-csv directory, basically making another copy of the Master.csv cdr file and also because I couldn't trust whether the Master.csv will be rotated accidentally again. Thanks On Fri, Dec 4, 2009 at 9:59 AM, Seven Du wrote: > 2009/12/4 Mouncifbb : > > I don't want to use XML cdr it puts each call on individual files so > > It posts to a http server, and fall back to a xml file if server fails > > > is it possible to include a JavaScript at the end of dialplan to > > collect info about the session? > > > > I think the answer is yes but where would you store the collected info? > > > Thanks > > > > > > On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > > > >> why not try mod_xml_cdr? > >> > >> 2009/12/4 Mouncif Benniane : > >>> is it possible to run a javascript at the end of dialplan to > >>> generate cdrs? > >>> because (mod_cdr_csv) is giving me hard time as it rotates Master > >>> file on > >>> machine reboots or shutdown signals. > >>> javascript or LUA for preferences? > >>> > >>> thank you > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/2ee358cd/attachment-0001.html From freeswitch-users-list at metik.com Fri Dec 4 07:24:36 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 04 Dec 2009 10:24:36 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <4B1929B4.5060507@metik.com> Since you seem to have most of the heavy lifting squared away with FS (e.g. database replication) and before reinventing the wheel, I would recommend that you speak to a few VoIP providers and see if they will do this for you as part of your service. Those that are using carrier class platforms (so-called active clustering) should be able to do this without too much effort on their part. If you reach any dead ends, please feel free to contact me off list. The one thing that you may want to keep in mind is that unless FS is not involved in the media flow (or has a chance to redirect the call to another FS), existing calls will be dropped. FS has no mechanism for exchanging/mirroring stateful signaling and media information between other FS nodes to specifically facilitate failover. As I believe the developers have indicated in this list, to do so would require significant investments in time and resources to implement it in the sofia sip stack at the moment. -metik Tim Uckun wrote: > I have read some of the archived emails about HA, loadbalancing, > failover etc and I am still a bit confused about how I could set up > some sort of resiliency with freeswitch. > > My situation is much less complex than the scenarios people were > talking about and I hoping the solution is similarly much less > complex. > > I have two machines. Both will run freeswitch and also an IVR > application with local databases. I will take care of the database, > application and configuration synchronization between the two > machines. Ideally the calls would be load balanced between the > machines and if any application falls down then the calls should go to > the other machine. Same if I take a machine down for whatever reason. > > If a machine goes down I am willing to "lose" those people who were > making a call at the time. I do have a flag in the application which > will stop answering the calls while processing the existing calls for > a graceful shutdown and hopefully the load balancer would shuttle the > calls to the other machine while this is happening. > > At this stage everything is done via SIP. > > My questions are... > > Do I have to have a sip proxy? If the answer is yes it seems like I > have to set up two sip proxies so I don't have another single point of > failure. Can I load the sip proxies on the same machine? Do I need two > more machines? > > If I take load balancing out of the picture would it be possible to do > a simple linux HA or a windows built in ip failover solution? Would a > simple IP failover work over UDP or would I have to use IAX and tcp/ip > ? > > Is it better to go the virtualization route? > > Sorry if these are dumb questions. I am just trying to get my head > wrapped around this. I don't need five nines (although that would be > awesome), I just want a reasonable degree of assurance that my app can > keep taking calls in case something weird happens. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Dec 4 07:38:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 09:38:59 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912040738r51056421t6122b3b408bc1be2@mail.gmail.com> That means you mysql is not configured to do transactions so it failed over back to sqlite. if you scan for the warning message you will see the option you have to set and you may possibly have to update your myodbc odbc driver. To answer you other question about the sqlite, like I said the lua does not have the object coded like js does so it would be a project to implement it. You can also consider using ODBC plugin for lua to access the sqlite. On Fri, Dec 4, 2009 at 3:24 AM, Jon Bruel wrote: > I have now tested the FS with core db configured using MySql (by > modifying the switch.conf.xml file). Unfortunately, it does not solve my > problem because some of the core tables still remain as active SQLite > tables. > > > > After restarting the FS in the new configuration (with SQLite database core > deleted), the following tables are created in MySql and SQLite: > > > > MySQL: aliases, complete, nat and tasks (database starting with no tables > prior to FS restart). > > SQLite: aliases, calls, channels, interfaces, nat and tasks. > > > > As I would like to access the channels table using Lua, the change did not > fix my problem. I have positive verified that the channels table is active > and populated during calls. > > > > Are there other places where I should define the usage of the MySql > database? > > > > > > *Jon Br?el* > Consiglia Telecommunications > > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > > CVR: 27047882 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/88f41a3b/attachment.html From neilp at cs.stanford.edu Fri Dec 4 07:58:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 21:28:58 +0530 Subject: [Freeswitch-users] IVR apps in lua Message-ID: Hi All, I haven't found a substantial example of IVR applications implemented in lua. Can anyone suggest where to look? My issue has to do with appropriate coding style. I am implementing a voice message board application in lua. I want to allow the user to dial buttons to navigate forward and back in the list of messages. One way to implement playmessage() is to check for a forward/back command while playing the current message, and if a command is given to invoke playmessage() with the prev/next message in the list. However, this leaves a chain of unreturned playmessage calls on the execution stack (a recursive function). Alternatively, the playmessage() function can return control to its caller (perhaps a while loop that spins forever) and pass back a code to indicate the command. The caller acts accordingly. This is non-recursive, but for anything but simple applications this style becomes tedious as you start needing to pass back more info and up longer chains of functions. Any guidance on this would be appreciated. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/392ebb7f/attachment.html From Prometheus001 at gmx.net Fri Dec 4 08:01:44 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 17:01:44 +0100 Subject: [Freeswitch-users] Voicmail - message only Message-ID: <4B193268.20009@gmx.net> Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter From nik.middleton at noblesolutions.co.uk Fri Dec 4 08:06:57 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 16:06:57 -0000 Subject: [Freeswitch-users] B Leg on bridged call is not hanging up Message-ID: Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint (read phone number) If the A leg hangs up, then the call is cleared down and all is well. However if the B Leg attempts to hang-up, the LUA script that is handling the bridge continues to play audio to the a leg, while the B leg is in limbo. It does eventually time out with no RTP. Running Sofia debug on the cli shows that I'm getting the BYE from the B Leg, but that's about as far as I can get. The hang-up hook is not being fired in the lua script. Anyone give me some pointers as to where I might start looking? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/06930b44/attachment.html From frank at carmickle.com Fri Dec 4 08:47:46 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 4 Dec 2009 11:47:46 -0500 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B193268.20009@gmx.net> References: <4B193268.20009@gmx.net> Message-ID: <20091204164746.GA31924@base.carmickle.com> Hello On Fri, Dec 04, Peter P GMX wrote: > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? Do you mean from the wiki http://wiki.freeswitch.org/wiki/Mod_voicemail#skip_instructions skip_instructions Skips playback of instructions when leaving messages. Variable is unset after voicemail application finishes. --FC From lists at redbonez.net Fri Dec 4 08:50:00 2009 From: lists at redbonez.net (Adam Ford) Date: Fri, 4 Dec 2009 09:50:00 -0700 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B193268.20009@gmx.net> References: <4B193268.20009@gmx.net> Message-ID: <01e801ca7501$d2b49eb0$781ddc10$@net> I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From frank at carmickle.com Fri Dec 4 09:19:27 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 4 Dec 2009 12:19:27 -0500 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <01e801ca7501$d2b49eb0$781ddc10$@net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> Message-ID: <20091204171927.GB31924@base.carmickle.com> On Fri, Dec 04, Adam Ford wrote: > I am still new to freeswitch, but I would think you could achieve this by > just passing the call to an IVR application that plays the message instead > of passing it to the voicemail application. > > -AF > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P > GMX > Sent: Friday, December 04, 2009 9:02 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Voicmail - message only > > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? Yeah. I guess it was unclear to me which part he wanted to switch off. You could just use playback or play_and_get_digits. --FC From Prometheus001 at gmx.net Fri Dec 4 09:26:14 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 18:26:14 +0100 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <01e801ca7501$d2b49eb0$781ddc10$@net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> Message-ID: <4B194636.7030306@gmx.net> I would like to manage this in the voicemail menu. "Press 6 to enable recording" "Press 7 to only play announcement" or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: > I am still new to freeswitch, but I would think you could achieve this by > just passing the call to an IVR application that plays the message instead > of passing it to the voicemail application. > > -AF > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P > GMX > Sent: Friday, December 04, 2009 9:02 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Voicmail - message only > > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? > > Best regards > Peter > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Fri Dec 4 09:33:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Dec 2009 09:33:32 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting! Message-ID: <87f2f3b90912040933h568df38ch87ca88c205d88e8f@mail.gmail.com> FYI, The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04 Please call in! :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/d981579b/attachment.html From lfurrea at gmail.com Fri Dec 4 10:16:14 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 4 Dec 2009 12:16:14 -0600 Subject: [Freeswitch-users] Sporadic call drops Message-ID: Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200 at 172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200 at 172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 200 at 172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/200 at 172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/5a512b1b/attachment.html From freeswitch at cartissolutions.com Fri Dec 4 11:32:56 2009 From: freeswitch at cartissolutions.com (Yossi Neiman) Date: Fri, 04 Dec 2009 13:32:56 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Message-ID: <4B1963E8.7050204@cartissolutions.com> A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as to troubleshoot a problem in FreeSWITCH. It's extremely frustrating to try to help people out who won't try the provided suggestions first. And note directly to "eaf" - bogomips is quite possibly the least significant bit of data about a cpu that you will get out of /proc/cpuinfo... The name itself - bogo, means bogus. http://en.wikipedia.org/wiki/Bogomips -Yossi From anthony.minessale at gmail.com Fri Dec 4 11:48:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 13:48:08 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <4B1963E8.7050204@cartissolutions.com> References: <26594250.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> Message-ID: <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> There is another user here with a 300mhz box. I am willing to investigate this improved performance for weak devices but I need to do it in a sane cross-platform way. On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman wrote: > A word to the wise to the general FreeSWITCH community: If Anthony > Minessale suggests that you try to do any number of things, it's a very > good idea to try all those ideas before continuing on. I've known him, > MikeJ, and bkw for several years, and they almost always have very good > ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > frustrating to try to help people out who won't try the provided > suggestions first. > > And note directly to "eaf" - bogomips is quite possibly the least > significant bit of data about a cpu that you will get out of > /proc/cpuinfo... The name itself - bogo, means bogus. > http://en.wikipedia.org/wiki/Bogomips > > -Yossi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/1b13c721/attachment.html From anthony.minessale at gmail.com Fri Dec 4 11:51:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 13:51:48 -0600 Subject: [Freeswitch-users] Sporadic call drops In-Reply-To: References: Message-ID: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> we changed that message a long time ago so people would not think that anymore We are now 3000 rev beyond the version you are at, I would like it if you try the lastest trunk. On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea wrote: > Hi all, > > Guys I know the question could be too vague, but I have a customer that > just reported frequent failure to place outbound calls though a PSTN gateway > on the LAN. > > I looked at the logs and I seem to be able to confirm that FS fails to > place the call through the gateway and that the issue resides on the FS side > since the first channel that s killed is tht of the internal extension > registered to FS and then FS send the BYE to gw and kills the channel. > > What are possible causes of this? > > I know you always like to look at complete logs but here's a snip that > could shed some light on the disconnection. (I can provide full logs if > required and worthed) > > 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/internal/200 at 172.16.3.5 entering state [ready][200] > 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/internal/200 at 172.16.3.5 entering state [terminated][200] > 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup > sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] > 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal sofia/internal/ > 200 at 172.16.3.5 [BREAK] > 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() > sofia/internal/200 at 172.16.3.5 ending bridge by request from write function > 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() > sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] > > > Is the 6th line normal behavior for ending the channel? > > FreeSWITCH Version 1.0.trunk (13484M) > > TIA > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bf8bf8fe/attachment-0001.html From anthony.minessale at gmail.com Fri Dec 4 12:00:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:00:25 -0600 Subject: [Freeswitch-users] B Leg on bridged call is not hanging up In-Reply-To: References: Message-ID: <191c3a030912041200k12c46c8dufe6573eac25bba43@mail.gmail.com> did you set the channel variable hangup_after_bridge=true on the A leg? On Fri, Dec 4, 2009 at 10:06 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > This one has me stumped. > > I'm originating a call, playing audio, trapping on DTMF and bridging to > another endpoint (read phone number) > > If the A leg hangs up, then the call is cleared down and all is well. > However if the B Leg attempts to hang-up, the LUA script that is handling > the bridge continues to play audio to the a leg, while the B leg is in > limbo. It does eventually time out with no RTP. > > Running Sofia debug on the cli shows that I'm getting the BYE from the B > Leg, but that's about as far as I can get. The hang-up hook is not being > fired in the lua script. > > Anyone give me some pointers as to where I might start looking? > > regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/23929090/attachment.html From anthony.minessale at gmail.com Fri Dec 4 12:01:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:01:50 -0600 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B194636.7030306@gmx.net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> <4B194636.7030306@gmx.net> Message-ID: <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX wrote: > I would like to manage this in the voicemail menu. > "Press 6 to enable recording" > "Press 7 to only play announcement" > or so. So hte user can manage it's settings on his own. > > Best regrds > Peter > > Adam Ford schrieb: > > I am still new to freeswitch, but I would think you could achieve this by > > just passing the call to an IVR application that plays the message > instead > > of passing it to the voicemail application. > > > > -AF > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Peter P > > GMX > > Sent: Friday, December 04, 2009 9:02 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Voicmail - message only > > > > Hello, > > > > is there a chance to have the voicemail system to play announcment #1 > > only and not play announcement and then record the voicemail? > > Means: Can I switch off the recording part? > > > > Best regards > > Peter > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/66a55479/attachment.html From nik.middleton at noblesolutions.co.uk Fri Dec 4 12:03:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 20:03:46 -0000 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge Message-ID: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/98eb7cbe/attachment.html From jerry.richards at teotech.com Fri Dec 4 12:47:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 4 Dec 2009 12:47:41 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Message-ID: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry From anthony.minessale at gmail.com Fri Dec 4 12:56:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:56:33 -0600 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge In-Reply-To: References: Message-ID: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi, > > > > Is there an option to hang-up both call legs in a bridge when one leg hangs > up? > > > > In my lua script I only ever see the hang-up for the call I?m in, not for > the bridged b leg. That said, I can see both a hang-up and un bridge event > being fired for the B leg. However my issue is that the A leg is still up, > and if I?ve called 2 Pots numbers, the phone network will maintain the > bridge. > > > > Is my only option to subscribe to the unbridge event and fire a hang-up > event using the ?other leg? UID? > > > > Regards, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/b3a7c9e0/attachment-0001.html From anthony.minessale at gmail.com Fri Dec 4 12:59:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:59:38 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> Message-ID: <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: > > I have Mediant 1000 gateway, and for some reason, when I make an outbound > call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A > Wireshark trace shows that FS is replying to the gateway's inbound RTP > packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP > packets to the same port that FS specified in the outbound INVITE. It > appears in the log that FS is discarding the 200 OK from the gateway. > > I disabled the Firewall and SELinux on the Freeswitch machine. I tried > changing to "true" and also "proxy", but it has no effect. > > Anyone know what could be the issue? I posted the Freeswitch log in the > pastebin. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/c1c71f9f/attachment.html From nik.middleton at noblesolutions.co.uk Fri Dec 4 13:16:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 21:16:58 -0000 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge In-Reply-To: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> References: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> Message-ID: Thanks for that, no didn't see the message, there seems to be a big delay in the messages getting turned around on the list. Yup, works great thanks. Script doesn't get events, so there was no way to check for the b leg hang-up. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 04 December 2009 20:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Option to hang-up both legs in a bridge did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton wrote: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/db24fb60/attachment.html From pjintheusa at gmail.com Fri Dec 4 13:29:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 4 Dec 2009 16:29:27 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system Message-ID: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/7d72dd95/attachment.html From anthony.minessale at gmail.com Fri Dec 4 14:16:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 16:16:00 -0600 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Message-ID: <191c3a030912041416t1345736fs5548e7ca889006bb@mail.gmail.com> you could make an endpoint module for FS that speaks the special protocol then use that to call the conference. On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones wrote: > Hi All, > > Every so often you have to ask a question - where you know so little - it's > hard to even now where to start. This is one of the times. I am not > expecting an full answer here, just a gentle nudge in right direction to get > me started. > > What I have is a propriety IP based conference system - who want to add the > ability to have inbound PSTN callers join their conferences. All their > signaling is propriety - no SIP - but I do have access to that signaling > schema so can do some translation. Enough to get the IP / Port & CODEC of > the RTP stream. They use speex rtp sessions over TCP. > > So from an architectural point of view I am thinking of having the callers > enter a FS conference and than bridge that conference to their IP based > conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a Sofia > gateway) to that IP based conference system I am kind of lost. But it seems > reasonable that I should be able to get my head round this, because I know > the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/5410219c/attachment-0001.html From mgg at giagnocavo.net Fri Dec 4 14:16:50 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 4 Dec 2009 17:16:50 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/47ecb171/attachment.html From kristian.kielhofner at gmail.com Fri Dec 4 14:41:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 4 Dec 2009 17:41:12 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Message-ID: <2d9149cd0912041441v5b5ef62bjf9bd5afe833f20a5@mail.gmail.com> A little more data from one of my (our) boxes: starbox_352 ~ # uname -a Linux starbox_352 2.6.26.8-astlinux #1 PREEMPT Tue Nov 24 16:20:52 EST 2009 i586 unknown starbox_352 ~ # starbox_352 ~ # cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 5 model : 10 model name : Geode(TM) Integrated Processor by AMD PCS stepping : 2 cpu MHz : 498.053 cache size : 128 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de pse tsc msr cx8 sep pge cmov clflush mmx mmxext 3dnowext 3dnow bogomips : 997.21 clflush size : 32 power management: starbox_352 ~ # cat /etc/astlinux-release astlinux-s2s-3491 starbox_352 ~ # I'll find one that has been in production for a while with some active calls... On Thu, Dec 3, 2009 at 6:49 PM, Anthony Minessale wrote: > Sigh, > > You just took it up a notch in terms of disdain and sarcasm. > Why do people always only apologize sarcastically? > > I asked you to try the -hp and turn off the monotonic clock just to gather > the results to help you.? You completely missed it and just went on about > the threads.?? Please save the "ok fine the code is perfect, blah blah" if > you would have just read the email and answered the question I might have > cared more about the status of your problem. > > I told you both of those threads need to be on their toes because they try > to balance between a certian number of sql stmts or 500ms whatever comes > first.? When there are thousands of events per second being turned into SQL > statements which are in turn compiled into large sql transactions. > > If you want to come up with a way that they can sleep longer until there is > a sign of activity and stay busy for a few seconds then slow down again, > that's probably possible but the process is already idle at 0% cpu so maybe > you can appreciate why we are not rushing to work on it.? Maybe I'll give it > a go just to show you it has nothing to do with your problem. > > Please don't mock our comment about several years.? You have no idea how > hard this code was to develop and it's truly insulting.? Its clear to see > you are locked into assuming that the busy threads that are not all that > busy because they are constantly yielding to the scheduler is breaking the > timing code.? I begged you to understand me when i told you that the err is > not normal, most boxes do not see it doing nothing and there has to be a > specific problem on your box or configuration.? So instead of working with > us you want to escalate to snotty comments.? That's pretty normal on the > internet I guess.....? If you want to have a constructive conversation about > our core, install FS on a normal box, use it for a few weeks, figure out > everything about how it works then try.... There was pure speculation and > conjecture in your original emails and I never said a word about it until > you kept pushing. > > Kristian mentioned he never sees that on that same hardware did you even > consider following up on why that is? > > I don't have your device, but I assume if you get it working well it will > certainly help you more than it helps me so you could at least have the > decency to believe what we are trying to tell you. > > > > > > > > On Thu, Dec 3, 2009 at 3:44 PM, eaf wrote: >> >> Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do >> that. >> At the moment, I hope it won't be necessary as I can make those "hyper" >> threads behave, and will see how that goes first. I see where your >> implementation could be coming from. There is a queue of SQL queries in >> sofia.c processed by the worker thread. There are only two pop functions >> available in APR: queue_pop() and queue_trypop(), so alas no option with a >> timeout here. You don't want to block the thread in pop() indefinitely >> because you chose that same worker needs to do ireg and gw processing once >> in a while (separated by tens or hundreds of seconds, btw). You also want >> to >> be able to detect shutdown condition so that the worker doesn't hold up >> profile thread. So you chose to poll for events every millisecond instead >> of >> just creating an apr_thread_cond_t for resource friendly signalling. >> >> I agree that the timer thread philosophy is great and was the right choice >> for scaling, but I just don't comprehend responses to things like these >> other SQL or sofia worker threads. Did somebody even remotely acknowledge >> that busy loops at least in those areas that I showed may probably be a >> bad >> idea and could've been eliminated? I've heard suggestions to bump up >> priority, I've heard that the code was perfect already, that it's the >> result >> of 4-year effort, that I am arrogant, don't listen and don't understand >> squat. >> >> I'm sorry if I gave you impression that I was looking for the bad parts in >> the software. I apologized for that already. All I wanted was to have >> constructive conversation, perhaps I'm not too good at it. Code is already >> perfect according to you? Fine with me. >> >> >> Anthony Minessale-2 wrote: >> > >> > no, >> > >> > I mean the one after that that you must have completely skipped with a >> > command line option to try and a param to set in the config. It somewhat >> > annoys me for taking the time to compose it now. ?I wrote all of the >> > code >> > you are talking about myself and I was trying to give you some >> > suggestions.... >> > >> > Well, actually, ?you did answer my question about the platform so you >> > must >> > have seen it..... >> > >> > The loops are not the cause of that migration message, something wrong >> > with >> > the hardware or the kernel is. >> > Another guy just told you he does not see that problem on the same exact >> > hardware. >> > >> > Even if you have a point about the sql threads, you could make a patch >> > to >> > slow them down but you cant slow down too much or you will not be able >> > to >> > handle 400 cps all asking to send updates to transactions in batches of >> > thousands of sql stmts. ?Every line of that code is carefully designed >> > so >> > I >> > don't know what else to tell you but to stop being so arrogant and >> > re-read >> > this thread for all the advice you have totally ignored. ?I started out >> > trying to help you but I have a lot of work to do. ?I thoroughly >> > explained >> > it to you and you are choosing to ignore me so I guess I'm done. >> > You can do whatever you want with your working copy, i'll see you in 3 >> > or >> > 4 >> > years when you get up to speed with the rest of us........ >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mayamatakeshi at gmail.com Fri Dec 4 14:45:17 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 5 Dec 2009 07:45:17 +0900 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: References: Message-ID: <15b9404e0912041445m19c5640avfc43bd17c960ea68@mail.gmail.com> I had this same problem today. I solved it using OPTION = 67108864 instead of OPTIONS = 67108864 I'm using CentOS5.3 (x86_64) br, takeshi On Sat, Nov 28, 2009 at 12:36 AM, Frank @ Impact wrote: > Yes. I am using version 5.1 I am using Fedora 12. > > > > -----Original Message----- > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Leon de > Rooij > *Sent:* Friday, November 27, 2009 10:19 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > > > Are you using the myodbc 3.51.18 version or higher ? > > > > I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > upgrade from jaunty.. > > > > regards, > > > > Leon > > > > > > On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > > > > Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > > > > [freeswitch] > > Driver = MySQL > > SERVER = 127.0.0.1 > > PORT = 4040 > > DATABASE = mydb > > OPTIONS = 67108864 > > > > ?I still get FS complaining with this. > > > > Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] > sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If > you are using mysql, make sure you are using MYODBC 3.51.18 or higher and > enable FLAG_MULTI_STATEMENTS > > > > FreeSWITCH>version > > FreeSWITCH Version 1.0.trunk (15660) > > > > Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST > 2009 x86_64 x86_64 x86_64 GNU/Linux > > > > From /etc/odbcinst.ini > > DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > > Setup = /usr/lib64/libodbcmyS.so > > > > Is this a FS issue ? or an issue with mysql odbc? Any insight would be > great. > > > > -----Original Message----- > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Leon de Rooij > *Sent:* Friday, November 27, 2009 3:37 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > > > There's a little info here on how to enable it with odbc: > > > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > > > regards, > > > > Leon > > > > > > On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > > > > > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: > > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html > > > > MySQL Connector/ODBC now supports batched statements. In order to enable > > cached statement support you must switch enable the batched > > statement option (FLAG_MULTI_STATEMENTS, > > 67108864, or Allow multiple statements > > within a GUI configuration). Be aware that batched statements > > create an increased chance of SQL injection attacks and you must > > ensure that your application protects against this scenario. > > (Bug#7445 ) > > > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/0fe3022d/attachment-0001.html From Prometheus001 at gmx.net Fri Dec 4 14:51:31 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 23:51:31 +0100 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> <4B194636.7030306@gmx.net> <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> Message-ID: <4B199273.6090301@gmx.net> Hello Anthony, thanks for the hint. I have posted a $100 bounty in the wiki + another $150 bounty to enable speaking an announcement via TTS. Best regards Peter Anthony Minessale schrieb: > You could file it as a feature request and post a bounty and probably > get the functionality fairly inexpensively maybe $100 > > > > On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX > wrote: > > I would like to manage this in the voicemail menu. > "Press 6 to enable recording" > "Press 7 to only play announcement" > or so. So hte user can manage it's settings on his own. > > Best regrds > Peter > > Adam Ford schrieb: > > I am still new to freeswitch, but I would think you could > achieve this by > > just passing the call to an IVR application that plays the > message instead > > of passing it to the voicemail application. > > > > -AF > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf > Of Peter P > > GMX > > Sent: Friday, December 04, 2009 9:02 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: [Freeswitch-users] Voicmail - message only > > > > Hello, > > > > is there a chance to have the voicemail system to play > announcment #1 > > only and not play announcement and then record the voicemail? > > Means: Can I switch off the recording part? > > > > Best regards > > Peter > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pjintheusa at gmail.com Fri Dec 4 14:58:52 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 4 Dec 2009 17:58:52 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> Message-ID: <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo wrote: > I think you will need to sort out the signaling first, as you?ll have to > tell the conference system to accept which RTP streams for which > conferences, as well as tell it to transmit to your callers, no? > > > > After that, then I would imagine you just need to do SDP rewriting when a > call hits FreeSWITCH. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Friday, December 04, 2009 2:29 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Bridging to a non SIP based system > > > > Hi All, > > Every so often you have to ask a question - where you know so little - it's > hard to even now where to start. This is one of the times. I am not > expecting an full answer here, just a gentle nudge in right direction to get > me started. > > What I have is a propriety IP based conference system - who want to add the > ability to have inbound PSTN callers join their conferences. All their > signaling is propriety - no SIP - but I do have access to that signaling > schema so can do some translation. Enough to get the IP / Port & CODEC of > the RTP stream. They use speex rtp sessions over TCP. > > So from an architectural point of view I am thinking of having the callers > enter a FS conference and than bridge that conference to their IP based > conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a Sofia > gateway) to that IP based conference system I am kind of lost. But it seems > reasonable that I should be able to get my head round this, because I know > the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/133343b9/attachment.html From anthony.minessale at gmail.com Fri Dec 4 15:41:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 17:41:19 -0600 Subject: [Freeswitch-users] Playing an rtp stream In-Reply-To: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> References: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> Message-ID: <191c3a030912041541j12732ef6t1577cf550c811375@mail.gmail.com> yes this is possible assuming that is a either a multicast address or a dedicated unicast address you want to listen on that something else is sending audio to. it would also require writing a module in C to actually implement it. On Thu, Dec 3, 2009 at 7:47 PM, Phillip Jones wrote: > Hi there, > > It it possible do something like: > > > > > > > > > > Basically I have need to connect to incoming calls listen to an existing > rtp stream - I know the IP and port. > > Any hints on achieving this would be much appreciated. > > Thanks > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/3076a59b/attachment.html From anthony.minessale at gmail.com Fri Dec 4 15:48:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 17:48:21 -0600 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: References: Message-ID: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> set rotate-on-hup to false in the cdr_csv config file then it will only rotate when the file gets too big and also you can get a cdr with session.generateXmlCdr() and dig out what you need or get it from variables but it will not be nearly as reliable as using the C ones because you need low level access to make sure you write to the disk properly from many threads etc. On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane wrote: > is it possible to run a javascript at the end of dialplan to generate cdrs? > because (mod_cdr_csv) is giving me hard time as it rotates Master file on > machine reboots or shutdown signals. > javascript or LUA for preferences? > > thank you > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/98c28996/attachment.html From msc at freeswitch.org Fri Dec 4 16:35:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Dec 2009 16:35:45 -0800 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: <87f2f3b90912041635s134375d0ife1b5a76e337f097@mail.gmail.com> Looks good so far. Try "oz list" and "oz dump 1" and see what happens. -MC On Thu, Dec 3, 2009 at 10:36 PM, Neil Patel wrote: > Thanks all for your help. I got around this by running ./Setup and > installing wanpipe in TDM API mode (it says it's the default for FS). I then > uncommented the mod_openzap line in modules.conf when installing FS. Finally > I ran wancfg_fs which creates appropriate config files for you for your FS > installation. I believe openzap is now installed properly: > > 2009-12-04 12:04:52.411017 [INFO] zap_io.c:2451 Loading IO from > /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] > 2009-12-04 12:04:52.411126 [INFO] zap_io.c:2251 auto-loaded 'wanpipe' > 2009-12-04 12:04:52.411311 [INFO] ozmod_wanpipe.c:287 configuring device > s1c1 as OpenZAP device 1:1 fd:14 DTMF: software > 2009-12-04 12:04:52.411377 [INFO] ozmod_wanpipe.c:287 configuring device > s1c2 as OpenZAP device 1:2 fd:15 DTMF: software > 2009-12-04 12:04:52.411444 [INFO] ozmod_wanpipe.c:287 configuring device > s1c3 as OpenZAP device 1:3 fd:17 DTMF: software > 2009-12-04 12:04:52.411509 [INFO] ozmod_wanpipe.c:287 configuring device > s1c4 as OpenZAP device 1:4 fd:18 DTMF: software > 2009-12-04 12:04:52.411575 [INFO] ozmod_wanpipe.c:287 configuring device > s1c5 as OpenZAP device 1:5 fd:19 DTMF: software > 2009-12-04 12:04:52.411639 [INFO] ozmod_wanpipe.c:287 configuring device > s1c6 as OpenZAP device 1:6 fd:20 DTMF: software > 2009-12-04 12:04:52.411707 [INFO] ozmod_wanpipe.c:287 configuring device > s1c7 as OpenZAP device 1:7 fd:21 DTMF: software > 2009-12-04 12:04:52.411771 [INFO] ozmod_wanpipe.c:287 configuring device > s1c8 as OpenZAP device 1:8 fd:22 DTMF: software > 2009-12-04 12:04:52.411837 [INFO] ozmod_wanpipe.c:287 configuring device > s1c9 as OpenZAP device 1:9 fd:23 DTMF: software > 2009-12-04 12:04:52.411903 [INFO] ozmod_wanpipe.c:287 configuring device > s1c10 as OpenZAP device 1:10 fd:24 DTMF: software > 2009-12-04 12:04:52.411969 [INFO] ozmod_wanpipe.c:287 configuring device > s1c11 as OpenZAP device 1:11 fd:25 DTMF: software > 2009-12-04 12:04:52.412034 [INFO] ozmod_wanpipe.c:287 configuring device > s1c12 as OpenZAP device 1:12 fd:26 DTMF: software > 2009-12-04 12:04:52.412102 [INFO] ozmod_wanpipe.c:287 configuring device > s1c13 as OpenZAP device 1:13 fd:27 DTMF: software > 2009-12-04 12:04:52.412179 [INFO] ozmod_wanpipe.c:287 configuring device > s1c14 as OpenZAP device 1:14 fd:28 DTMF: software > 2009-12-04 12:04:52.412244 [INFO] ozmod_wanpipe.c:287 configuring device > s1c15 as OpenZAP device 1:15 fd:29 DTMF: software > TDM API: CMD: 18 > : Operation not supported > 2009-12-04 12:04:52.412416 [INFO] ozmod_wanpipe.c:287 configuring device > s1c16 as OpenZAP device 1:16 fd:30 DTMF: none > 2009-12-04 12:04:52.412503 [INFO] ozmod_wanpipe.c:287 configuring device > s1c17 as OpenZAP device 1:17 fd:31 DTMF: software > 2009-12-04 12:04:52.412568 [INFO] ozmod_wanpipe.c:287 configuring device > s1c18 as OpenZAP device 1:18 fd:32 DTMF: software > 2009-12-04 12:04:52.412634 [INFO] ozmod_wanpipe.c:287 configuring device > s1c19 as OpenZAP device 1:19 fd:33 DTMF: software > 2009-12-04 12:04:52.412708 [INFO] ozmod_wanpipe.c:287 configuring device > s1c20 as OpenZAP device 1:20 fd:34 DTMF: software > 2009-12-04 12:04:52.412771 [INFO] ozmod_wanpipe.c:287 configuring device > s1c21 as OpenZAP device 1:21 fd:35 DTMF: software > 2009-12-04 12:04:52.412838 [INFO] ozmod_wanpipe.c:287 configuring device > s1c22 as OpenZAP device 1:22 fd:36 DTMF: software > 2009-12-04 12:04:52.412902 [INFO] ozmod_wanpipe.c:287 configuring device > s1c23 as OpenZAP device 1:23 fd:37 DTMF: software > 2009-12-04 12:04:52.412948 [INFO] ozmod_wanpipe.c:287 configuring device > s1c24 as OpenZAP device 1:24 fd:38 DTMF: software > 2009-12-04 12:04:52.412988 [INFO] ozmod_wanpipe.c:287 configuring device > s1c25 as OpenZAP device 1:25 fd:39 DTMF: software > 2009-12-04 12:04:52.413018 [INFO] ozmod_wanpipe.c:287 configuring device > s1c26 as OpenZAP device 1:26 fd:40 DTMF: software > 2009-12-04 12:04:52.413041 [INFO] ozmod_wanpipe.c:287 configuring device > s1c27 as OpenZAP device 1:27 fd:41 DTMF: software > 2009-12-04 12:04:52.413063 [INFO] ozmod_wanpipe.c:287 configuring device > s1c28 as OpenZAP device 1:28 fd:42 DTMF: software > 2009-12-04 12:04:52.413086 [INFO] ozmod_wanpipe.c:287 configuring device > s1c29 as OpenZAP device 1:29 fd:43 DTMF: software > 2009-12-04 12:04:52.413106 [INFO] ozmod_wanpipe.c:287 configuring device > s1c30 as OpenZAP device 1:30 fd:44 DTMF: software > 2009-12-04 12:04:52.413128 [INFO] ozmod_wanpipe.c:287 configuring device > s1c31 as OpenZAP device 1:31 fd:45 DTMF: software > 2009-12-04 12:04:52.413142 [INFO] zap_io.c:2374 Configured 31 channel(s) > 2009-12-04 12:04:52.431405 [INFO] zap_io.c:2468 Loading SIG from > /usr/local/freeswitch/mod/ozmod_ss7_boost.so > 2009-12-04 12:04:52.431441 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost' > 2009-12-04 12:04:52.431541 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [mod_openzap] > 2009-12-04 12:04:52.431553 [NOTICE] switch_loadable_module.c:142 Adding > Endpoint 'openzap' > 2009-12-04 12:04:52.431638 [NOTICE] switch_loadable_module.c:248 Adding > Application 'disable_ec' > 2009-12-04 12:04:52.431659 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'oz' > 2009-12-04 12:04:52.432009 [WARNING] ss7_boost_client.c:244 TX EVENT (P): > SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0 > > > Does this look right? > Thanks. > > On Mon, Nov 30, 2009 at 9:09 PM, Moises Silva wrote: > >> On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: >> >>> Hi All, >>> >>> I am currently installing a Sangoma A102 card to work with FS using >>> wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get >>> openzap-related modules to compile: >>> >>> > cd wanpipe-3.5.6.5/ >>> > make openzap >>> ... >>> make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >>> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >>> make -C api/libstelephony clean >>> make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >>> make[1]: *** No rule to make target `clean'. Stop. >>> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >>> make: *** [all_lib] Error 2 >>> >>> The libstelephony directory has no Makefile in it. Why is it missing? Is >>> there a version of wanpipe drivers that will work? I have been unsuccessful >>> with 3.4.4 and 3.5.6 in similar fashion. >>> >>> >> Hi Neil, >> >> Most likely the creation of the Makefile failed (since you mention you >> can't see a Makefile). Please be sure to have installed the pre-requisites >> listed at http://wiki.sangoma.com/Requirements >> >> Particularly in this case, libtool, autoconf and automake packages. >> >> -- >> Moises Silva >> Software Developer >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >> 9T3 Canada >> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/46fb4fb7/attachment-0001.html From mgg at giagnocavo.net Fri Dec 4 17:52:59 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 4 Dec 2009 20:52:59 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you're going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it's just me ;) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo > wrote: I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/f153580f/attachment.html From mrene_lists at avgs.ca Fri Dec 4 18:02:21 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 21:02:21 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> Message-ID: <3943EA1B-27CA-4400-96C4-CB6FD344B916@avgs.ca> You can re-use some of mod_sofia's functions (like sofia_glue_parse_sdp) and only write the part of signalling thats different from SIP. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 8:52 PM, Michael Giagnocavo wrote: > Yes I was just thinking that it might be simpler to just fixup the > SDP and just write some custom script to talk control to the backend > conference system than to write a whole endpoint module. Especially > cause you can do the fixup and control in a high level language > (even if you use C#, you?re going to end up playing with pointers > except the syntax will be more verbose). Then again, I have a > natural aversion to C so maybe it?s just me ;) > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Friday, December 04, 2009 3:59 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Bridging to a non SIP based system > > Ah guys - that was exactly the nudge I was looking for - I will take > a look at the other endpoint modules like mod_skypiax etc. I will > also look at the SDP - I see where you are going there - I might not > even need the conference in that case. > > Question is - could I write an endpoint is C# !!! :) > > Thanks again - that's a great help. > > On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo > wrote: > I think you will need to sort out the signaling first, as you?ll > have to tell the conference system to accept which RTP streams for > which conferences, as well as tell it to transmit to your callers, no? > > After that, then I would imagine you just need to do SDP rewriting > when a call hits FreeSWITCH. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Friday, December 04, 2009 2:29 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Bridging to a non SIP based system > > Hi All, > > Every so often you have to ask a question - where you know so little > - it's hard to even now where to start. This is one of the times. I > am not expecting an full answer here, just a gentle nudge in right > direction to get me started. > > What I have is a propriety IP based conference system - who want to > add the ability to have inbound PSTN callers join their conferences. > All their signaling is propriety - no SIP - but I do have access to > that signaling schema so can do some translation. Enough to get the > IP / Port & CODEC of the RTP stream. They use speex rtp sessions > over TCP. > > So from an architectural point of view I am thinking of having the > callers enter a FS conference and than bridge that conference to > their IP based conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a > Sofia gateway) to that IP based conference system I am kind of lost. > But it seems reasonable that I should be able to get my head round > this, because I know the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/b98b7ef3/attachment-0001.html From andrewkt at aktzero.com Fri Dec 4 18:40:12 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 04 Dec 2009 21:40:12 -0500 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03c401ca73bf$1cea8600$56bf9200$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: <4B19C80C.2060508@aktzero.com> On 12/2/2009 9:19 PM, Lars Zeb wrote: > Is this reasonable given it was the only call in FreeSwitch at the time? How > can this situation be corrected in the future? As a workaround, you can eavesdrop with 779, and use * to navigate channels. -- Andrew Thompson From pmhshz at gmail.com Fri Dec 4 23:17:36 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 4 Dec 2009 23:17:36 -0800 (PST) Subject: [Freeswitch-users] Need Conference design help Message-ID: <26653473.post@talk.nabble.com> Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3. User who can only participate when Moderator allow them to get in. Also besides above setup I have to perform other things like Record the conference, Multicast the conference to other freeswitch server. I saw the conference Record CLI command but wondering where to setup when conference starts. I am also wondering how Multicast Conference is possible in Freeswitch and how the receiver Freeswitch configuration will look like. Thanks. msp -- View this message in context: http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rsavage at KingBallow.com Fri Dec 4 12:07:14 2009 From: rsavage at KingBallow.com (Reece Savage) Date: Fri, 4 Dec 2009 14:07:14 -0600 Subject: [Freeswitch-users] Aastra XML scripts. Message-ID: <8E4ACA7747F7F641991455BC157390C80145007D@srv-nash-ex.mail.kingballow.com> Would anyone be willing to port the Aastra XML scripts for Asterisk to FreeSWITCH? I would be willing to sponser. Reece Savage Information Technology Manager King & Ballow Law Offices 315 Union Street Suite 1100 Nashville, TN 37201 Phone (615) 726-5525 Fax (615) 254-7907 rsavage at kingballow.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/fcfad003/attachment.html From mailinglist at fribert.dk Sat Dec 5 01:46:30 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sat, 05 Dec 2009 10:46:30 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? Message-ID: <4B1A3A06020000E100000289@mail.fribert.dk> Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/c260b101/attachment.html From testeador01 at gmail.com Sat Dec 5 07:02:00 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 5 Dec 2009 10:02:00 -0500 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: Hello and welcome to FreeSWITCH, This is the starter's guide: http://wiki.freeswitch.org/wiki/Getting_Started_Guide Also Michael Collins wrote this nice article that will help you get started in VoIP and Freeswitch: http://bit.ly/EpVrv Most of the FreeSWITCH features are documented in the wiki, although I suggest not searching in the wiki search box but using google. 2009/12/5 mailinglist > Has anybody done this? > > I'm completely at a loss, having tinkered very little with Asterisk, and > giving up on that, I wonder if there's any help to be found on FreeSwitch? > Anybody that can give pointers to a good step-by-step instruction? > > I want to have it handle my two sip-phones (siemens dect ip and spa 901), > and handle a sip account at my provider. > Of course transferring calls between the two, as well as group calls would > be a nice benefit. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/5130f644/attachment.html From talk2ram at gmail.com Sat Dec 5 07:22:56 2009 From: talk2ram at gmail.com (ram) Date: Sat, 5 Dec 2009 20:52:56 +0530 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: On Sat, Dec 5, 2009 at 3:16 PM, mailinglist wrote: > Has anybody done this? > > I'm completely at a loss, having tinkered very little with Asterisk, and > giving up on that, I wonder if there's any help to be found on FreeSwitch? > Anybody that can give pointers to a good step-by-step instruction? > > I want to have it handle my two sip-phones (siemens dect ip and spa 901), > and handle a sip account at my provider. > Of course transferring calls between the two, as well as group calls would > be a nice benefit. > > > in short answer Fusionpbx.com Ram > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/67b628b5/attachment.html From nik.middleton at noblesolutions.co.uk Sat Dec 5 07:41:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 15:41:34 -0000 Subject: [Freeswitch-users] how to disable hook flash hold Message-ID: Hi, Is it possible to disable being able to put a call on hold using hook flash? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/b14a698f/attachment.html From testeador01 at gmail.com Sat Dec 5 08:05:02 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 5 Dec 2009 11:05:02 -0500 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi,? Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pjintheusa at gmail.com Sat Dec 5 08:49:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Sat, 5 Dec 2009 11:49:27 -0500 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: <367751820912050849s2e0acd45we4c60989af78f8bd@mail.gmail.com> Also check out this great write up: http://wiki.freeswitch.org/wiki/Multi_home_tutorial This is pfSense specific. On Sat, Dec 5, 2009 at 10:22 AM, ram wrote: > > > On Sat, Dec 5, 2009 at 3:16 PM, mailinglist wrote: > >> Has anybody done this? >> >> I'm completely at a loss, having tinkered very little with Asterisk, and >> giving up on that, I wonder if there's any help to be found on FreeSwitch? >> Anybody that can give pointers to a good step-by-step instruction? >> >> I want to have it handle my two sip-phones (siemens dect ip and spa 901), >> and handle a sip account at my provider. >> Of course transferring calls between the two, as well as group calls would >> be a nice benefit. >> >> >> > > in short answer Fusionpbx.com > > Ram > > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/c5c8c9f3/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sat Dec 5 09:20:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 17:20:33 -0000 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: Sorry, I meant from a POTS phone Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: 05 December 2009 16:05 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi,? Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lists at redbonez.net Sat Dec 5 09:52:37 2009 From: lists at redbonez.net (Adam Ford) Date: Sat, 5 Dec 2009 10:52:37 -0700 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: <0E0013F55E224674A1361329CF7A85F0@redbonez> I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office. http://doc.pfsense.org/index.php/FreeSWITCH -AF _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mailinglist Sent: Saturday, December 05, 2009 2:47 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/80f712b1/attachment.html From tculjaga at gmail.com Sat Dec 5 11:01:10 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 5 Dec 2009 20:01:10 +0100 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> The POTS phone is attached to something... (ZAP channel or an ATA or a gateway). It is there you configure this behaviour. T. On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Sorry, I meant from a POTS phone > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena > Sent: 05 December 2009 16:05 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] how to disable hook flash hold > > It can be done from the phone itself; for example on a Grandstream > phone it is done with the option "Onhook Threshold:" setting it to > "hookflash OFF" > > > 2009/12/5 Nik Middleton > > > > Hi, Is it possible to disable being able to put a call on hold using > hook flash? > > > > > > > > Regards > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/ecded617/attachment.html From pekkis50 at gmail.com Sat Dec 5 11:06:19 2009 From: pekkis50 at gmail.com (Pekka Kurki) Date: Sat, 05 Dec 2009 20:06:19 +0100 Subject: [Freeswitch-users] freeswitch binaries w/o IPv6 anywhere (for w2k)? Message-ID: <4B1AAF2B.1070305@gmail.com> thanks, --pekka-- From nik.middleton at noblesolutions.co.uk Sat Dec 5 11:32:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 19:32:51 -0000 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> References: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> Message-ID: It's a pots phone at the end of a VoIP trunk provided by my ISP. I have not control over it. The only think I have found so far is: Which is what I presume I add to my provider's conf file. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: 05 December 2009 19:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold The POTS phone is attached to something... (ZAP channel or an ATA or a gateway). It is there you configure this behaviour. T. On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton wrote: Sorry, I meant from a POTS phone Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: 05 December 2009 16:05 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi, Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/af67cd72/attachment-0001.html From Prometheus001 at gmx.net Sat Dec 5 12:35:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 05 Dec 2009 21:35:28 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> Message-ID: <4B1AC410.9050201@gmx.net> Hello Anthony, I did some checks today Here is how the phones are registered: mysql> select sip_host, presence_hosts, server_user,server_host, hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip_host | presence_hosts | server_user | server_host | hostname | sip_realm | mwi_user | mwi_host | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ IPs are: 10.11.12.1 sip1.mydomain.com (common cluster IP) 10.11.12.2 sip11.mydomain.com 10.11.12.3 sip12.mydomain.com (not used at this point) XML-Curl for the directory is:
The internal profile has the following alias: With $${domain} being sip11.mydomain.com Phones are registering to sip1.mydomain.com, Voicemail works, but MWI does not. Any hint what I should change to make this work? Best regards Peter Anthony Minessale schrieb: > based on your example past > > sip1.mydomain.com is the domain in the > packet and thus the profile should have an alias for this. > Then the user must reside in your sip db with the user 200 and domain > sip1.mydomain.com > > if you dont have this consider the force-register-domain and > force-register-db-domain to normalize the host names. > > > On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale > > wrote: > > Did you check the 2 replies that told you you need aliases in your > sofia profile to translate the domain found in your > message_waiting to the right profile? Both Brian and Mike > answered you. > > > > > > On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX > > wrote: > > I tried now with phones directly attached to the freeswitch > (without an > OpenSIPS in between). I also added the alias. But the > behaviour is as > before: > No notify message from freeswitch, neither after register nor > after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: > > Yes an alias will be required for every domain you run on > the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > > > >> Try an alias on the sip profile. > >> > >> Mike > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Mailings at kh-dev.de Sat Dec 5 19:04:26 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 04:04:26 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Message-ID: Hi, currently I'm testing the newest FS trunk. Now I need a hint how to set up an "old" behavior of version 1.0.4. Here's the scenario: - Incoming call from caller_id_name: abc and caller_id_number: 123 - Now I set effective_caller_id_name: xyz and effective_caller_id_number: 456 - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) - After the pickup the Leg B is switching back to the "old" values and displays abc + 123 But I would rather see the new values during the call (as it was in version 1.0.4). What do I need to change? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/2ebc74d4/attachment.html From yehavi.bourvine at gmail.com Sun Dec 6 00:12:35 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 10:12:35 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Message-ID: Hello Ognjen, From the tests I've done it is not so... When I set the profile to use INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the FreeSwich ignores it (does not have phone-events field in the reply SDP) which causes the phone to not send RFC2833 events... Regards, __Yehavi: 2009/12/3 Ognjen Seslija > Bear in mind that FS will accept both 2833 and INFO in any profile on an > inbound call. Param "dtmf-type" is valid only for outbound calls from the > profile. > > Ognjen > > On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Hello, >> >> I have Polycom phones which send only RFC-2833 (or inband which I >> dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco >> gateway has some bug and accepts only INFO. >> >> I did a few tests: >> >> - Some of the phones are on different profile than the Cisco. On their >> profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set >> 'dtmf-type=info' and Freeswitch did the translation. All works ok... >> - Some of the phones are on the same profile as the Cisco, so I must >> set dtmf-type to rfc2833; it works with internal applications (like >> voicemail) but does not work through the Cisco as it misinterprets the >> rfc2833 >> >> >> Is there a way to set some variable (or a parameter to the bridge >> application) to do the translation? >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/03f58458/attachment.html From mailinglist at fribert.dk Sun Dec 6 01:24:49 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 10:24:49 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Message-ID: <4B1B8671020000E100000293@mail.fribert.dk> Hi Adam Excellent first steps! Thankyou for the hint. Now I hope somebody can tell me what I'm doing wrong next... I've gotten it to register to the testprovider here (musimi.dk), but I get an error when I create an account for testing with the X-Lite phone. It displays 403 forbidden in the display. I've created an account on FreeSwitch extension 1001 password 1001 mailbox 1001 voicemail password 1001 account code 1001 Effective Caller ID Name Fribert Effective Caller ID Number 4692xxxx (the Musimi number) Voicemail Mail To Voicemail Attach File true User Context default Call Group <> Enabled true Extension Description Test number In the X-Lite Display Name Fribert User name 1001 Password 1001 Autorization user name 1001 Domain LAN-IP-OF-pfSense Check in Register with domain and receive incoming calls Check in domain. That's about it. Looking on the status page, I can see these lines in the log: 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding 192.168.42.0/24 (deny) to list lan 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding 192.168.42.42/32 (allow) to list lan 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 Which I'm kinda confused about, I don't have any 192.168 net here??? But as it also primarily forbids it, except .42 to allow, I'm wondering if it could be something internal? Best regards >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office. http://doc.pfsense.org/index.php/FreeSWITCH -AF From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mailinglist Sent: Saturday, December 05, 2009 2:47 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/42fd5f2d/attachment-0001.html From freeswitch-users-list at metik.com Sun Dec 6 02:24:26 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 06 Dec 2009 05:24:26 -0500 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Message-ID: <4B1B865A.2060901@metik.com> You previously stated that your Cisco gateway has some "bug" that prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on the voip dial-peer that the call is using? Unless you have configured the Cisco to support assymetric SDP or are using a non-default "rtp payload-type nte" setting that does not agree to well with FS's (default) "rfc2833-pt" setting, you should not have to use (SIP) INFO unless you want to. I would recommend doing the following to ensure you are hitting the correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) output: DTMF Relay = enabled, dtmf-relay = rtp-nte, example: show dialplan number 5551212 | i (dtmf-relay|DTMF Relay) DTMF Relay = enabled, dtmf-relay = rtp-nte, Also, you can sift through "show sip-ua calls" for the call and ensure that the value of "Negotiated Dtmf-relay" is "rtp-nte". -metik Yehavi Bourvine wrote: > Hello Ognjen, > > From the tests I've done it is not so... When I set the profile to > use INFO, and a phone calls and asks for RFC2833 (phone-events in the > SDP) the FreeSwich ignores it (does not have phone-events field in the > reply SDP) which causes the phone to not send RFC2833 events... > > Regards, __Yehavi: > > 2009/12/3 Ognjen Seslija > > > Bear in mind that FS will accept both 2833 and INFO in any profile > on an inbound call. Param "dtmf-type" is valid only for outbound > calls from the profile. > > Ognjen > > On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine > > wrote: > > Hello, > > I have Polycom phones which send only RFC-2833 (or inband > which I dislike) and they should go out to the PSTN via a > Cisco gateway. The Cisco gateway has some bug and accepts only > INFO. > > I did a few tests: > > * > Some of the phones are on different profile than the > Cisco. On their profile I set 'dtmf-type=rfc2833' and on > the Cisco's profile I set 'dtmf-type=info' and > Freeswitch did the translation. All works ok... > * > Some of the phones are on the same profile as the Cisco, > so I must set dtmf-type to rfc2833; it works with > internal applications (like voicemail) but does not work > through the Cisco as it misinterprets the rfc2833 > > > Is there a way to set some variable (or a parameter to the > bridge application) to do the translation? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Sun Dec 6 03:59:01 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 13:59:01 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4B1B865A.2060901@metik.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> Message-ID: Hello Metik, 2009/12/6 Metik > You previously stated that your Cisco gateway has some "bug" that > prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on > the voip dial-peer that the call is using? > > It is a PSTN dialpeer here, and it cannot be defined on it... > Unless you have configured the Cisco to support assymetric SDP or are > using a non-default "rtp payload-type nte" setting that does not agree > to well with FS's (default) "rfc2833-pt" setting, you should not have to > use (SIP) INFO unless you want to. > > I would recommend doing the following to ensure you are hitting the > correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): > > command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) > > Unfortunately this does not work on PSTN dial peers. > > Also, you can sift through "show sip-ua calls" for the call and ensure > that the value of "Negotiated Dtmf-relay" is "rtp-nte". > > This indeed shows that it has negotiated rtp-nte. Even when I do debug for CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them while it accepts them via INFO. As I said: I guess this is a bug. Since the gateway is on a remote site I hesitate on upgrading it until I hae the chance to go there. Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/356baed7/attachment.html From Prometheus001 at gmx.net Sun Dec 6 06:14:42 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 15:14:42 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1B8671020000E100000293@mail.fribert.dk> References: <4B1B8671020000E100000293@mail.fribert.dk> Message-ID: <4B1BBC52.3050106@gmx.net> Concerning, > Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests. Can you set the console_log_level to "debug" in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb: > Hi Adam > > Excellent first steps! > Thankyou for the hint. > Now I hope somebody can tell me what I'm doing wrong next... > > I've gotten it to register to the testprovider here (musimi.dk), but I > get an error when I create an account for testing with the X-Lite phone. > > It displays 403 forbidden in the display. > > I've created an account on FreeSwitch > > extension 1001 > password 1001 > mailbox 1001 > voicemail password 1001 > account code 1001 > Effective Caller ID Name Fribert > Effective Caller ID Number 4692xxxx (the Musimi number) > Voicemail Mail To > Voicemail Attach File true > User Context default > Call Group <> > Enabled true > Extension Description Test number > > In the X-Lite > Display Name Fribert > User name 1001 > Password 1001 > Autorization user name 1001 > Domain LAN-IP-OF-pfSense > > Check in Register with domain and receive incoming calls > Check in domain. > > That's about it. > > Looking on the status page, I can see these lines in the log: > 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding > 192.168.42.0/24 (deny) to list lan > 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding > 192.168.42.42/32 (allow) to list lan > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > > Which I'm kinda confused about, I don't have any 192.168 net here??? > But as it also primarily forbids it, except .42 to allow, I'm > wondering if it could be something internal? > > Best regards > > > >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i > meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: > > I used the pfSense FreeSWITCH for awhile, as it is the only GUI > FreeSWITCH I have found with a stable release. It was very easy to > use, I would recommend it if you just want a quick base system with > standard features. Though, I ended up switching to a compiled version > of FreeSWITCH in order to make the customizations I needed for my office. > > http://doc.pfsense.org/index.php/FreeSWITCH > > -AF > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *mailinglist > *Sent:* Saturday, December 05, 2009 2:47 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a > pfSensefirewall??? > > > > Has anybody done this? > > > > I'm completely at a loss, having tinkered very little with Asterisk, > and giving up on that, I wonder if there's any help to be found on > FreeSwitch? > > Anybody that can give pointers to a good step-by-step instruction? > > > > I want to have it handle my two sip-phones (siemens dect ip and spa > 901), and handle a sip account at my provider. > > Of course transferring calls between the two, as well as group calls > would be a nice benefit. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jbr at consiglia.dk Sun Dec 6 06:22:41 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 6 Dec 2009 15:22:41 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: Dear all Some feedback regarding using Lua to access core database: First of all, I did not succeed to get SQLite drivers in Lua or ODBC-drivers in Lua to work. The SQLite driver did compile OK, but there was an error when loading into Lua. The ODBC driver did also compile OK, did load into Lua, but could not connect. Accessing the SQLite from the Linux console using "isql -v " worked OK. The problems may be related to the present Linux distribution, which is Ubuntu 9.1 server. Unfortunately the public searchable information about Lua ODBC driver problems is sparse. So I continued to try to get the FS to use MySQL as the core db. A number of problem occurred, which I did not find solution for in the FS documents. The problems and solutions are described below: 1) The core database is not automatically created by FS, therefore I created it manually. 2) During startup, the FS test for transaction support, and this test failed. To achieved transaction support with MySQL and MyODBC, three things had to be changed: a. A line was added in my.cnf to force innoDB as the default table: under the [mysqld] header, the following line was added: set-variable = default-table-type=InnoDB. b. The a line under the DNS was added to allow for multiple line statement support: option = 67108864. (ODBC version is 3.51). 3) After these changes the transaction worked, but all the tables in the core db were not created, therefore I copied the structure from the SQLite tables into tables with the same names in the MySQL database. This exercise also showed what the problem was: MySQL could not create tables with many VARCHAR type files with a size of 4096 (sound very big?). The size was reduced to 255, and most of the tables were created OK. One table still gave problems: the interface table. One of the fields is called key, which is a reserved word in MySQL, and by backticking the word key in the create statement, it worked. 4) Finally the FS started up using the MySQL, but errors splashed over the screen just after startup. There was a problem creating new records in the interface table, the problem was the key field. Changing the insert statement in switch.core.sqldb.c file by backticking the key field name and recompiling the FS solved that problem. I guess this will be fixed in later releases and I hope this will assist the brave programmers! I would like to argue for the development of SQLite connectivity in Lua. The ODBC core solution is not as clean as a direct database connection, and as long as this is limited to SQLite, a direct connection from "recommended script language" would be the cleanest solution. Further, it would be nice if everything works after having compiled the FS package. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/5c1f6c40/attachment-0001.html From Prometheus001 at gmx.net Sun Dec 6 08:22:30 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 17:22:30 +0100 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? Message-ID: <4B1BDA46.7010803@gmx.net> Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter From mailinglist at fribert.dk Sun Dec 6 10:13:02 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 19:13:02 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? Message-ID: <4B1C023E020000E1000002A2@mail.fribert.dk> Hi Peter Ok, I got the net changed in acl.conf.xml. I then tried setting console_loglevel, but I don't see any output on the console, it could very well be because it's a FreeBSD, and has very limited console. But after a restart it registers! So for some reason it needed a nudge there, very interesting. So now I have a local extension registered, and a provider registered, now I just need them to communicate. As I understand it, that's the dialplan I have to look at. I only have one provider hooked up, so a dial should be simple, right? It as Order 001 Condition ^0(.\d+)$ action bridge As I understand it, it should react on the 0 (for outside dialing) and then strip it And the action bridges the call to the outside. But I guess I'm missing something, because I just get a 'temporarily unavailable' shown in the xlite. >>> 06-12-2009 kl. 15:14 skrev Peter P GMX i meddelelsen <4B1BBC52.3050106 at gmx.net>: Concerning, > Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests. Can you set the console_log_level to "debug" in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb: > Hi Adam > > Excellent first steps! > Thankyou for the hint. > Now I hope somebody can tell me what I'm doing wrong next... > > I've gotten it to register to the testprovider here (musimi.dk), but I > get an error when I create an account for testing with the X-Lite phone. > > It displays 403 forbidden in the display. > > I've created an account on FreeSwitch > > extension 1001 > password 1001 > mailbox 1001 > voicemail password 1001 > account code 1001 > Effective Caller ID Name Fribert > Effective Caller ID Number 4692xxxx (the Musimi number) > Voicemail Mail To > Voicemail Attach File true > User Context default > Call Group <> > Enabled true > Extension Description Test number > > In the X-Lite > Display Name Fribert > User name 1001 > Password 1001 > Autorization user name 1001 > Domain LAN-IP-OF-pfSense > > Check in Register with domain and receive incoming calls > Check in domain. > > That's about it. > > Looking on the status page, I can see these lines in the log: > 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding > 192.168.42.0/24 (deny) to list lan > 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding > 192.168.42.42/32 (allow) to list lan > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > > Which I'm kinda confused about, I don't have any 192.168 net here??? > But as it also primarily forbids it, except .42 to allow, I'm > wondering if it could be something internal? > > Best regards > > > >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i > meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: > > I used the pfSense FreeSWITCH for awhile, as it is the only GUI > FreeSWITCH I have found with a stable release. It was very easy to > use, I would recommend it if you just want a quick base system with > standard features. Though, I ended up switching to a compiled version > of FreeSWITCH in order to make the customizations I needed for my office. > > http://doc.pfsense.org/index.php/FreeSWITCH > > -AF > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *mailinglist > *Sent:* Saturday, December 05, 2009 2:47 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a > pfSensefirewall??? > > > > Has anybody done this? > > > > I'm completely at a loss, having tinkered very little with Asterisk, > and giving up on that, I wonder if there's any help to be found on > FreeSwitch? > > Anybody that can give pointers to a good step-by-step instruction? > > > > I want to have it handle my two sip-phones (siemens dect ip and spa > 901), and handle a sip account at my provider. > > Of course transferring calls between the two, as well as group calls > would be a nice benefit. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/f46a2acd/attachment.html From lon at kickasspixels.com Sun Dec 6 10:13:01 2009 From: lon at kickasspixels.com (Lon Baker) Date: Sun, 6 Dec 2009 10:13:01 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> Jon, What version of MySQL are you using? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/d2f985d1/attachment.html From Mailings at kh-dev.de Sun Dec 6 10:38:54 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 19:38:54 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: References: Message-ID: Hi, I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone with caller_id_name and caller_id_number instead of effective_caller_id_name and effective_caller_id_number values. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Klaus Hochlehnert Sent: Sunday, December 06, 2009 4:04 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Hi, currently I'm testing the newest FS trunk. Now I need a hint how to set up an "old" behavior of version 1.0.4. Here's the scenario: - Incoming call from caller_id_name: abc and caller_id_number: 123 - Now I set effective_caller_id_name: xyz and effective_caller_id_number: 456 - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) - After the pickup the Leg B is switching back to the "old" values and displays abc + 123 But I would rather see the new values during the call (as it was in version 1.0.4). What do I need to change? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/a889bad5/attachment.html From mrene_lists at avgs.ca Sun Dec 6 10:42:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 6 Dec 2009 13:42:17 -0500 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: References: Message-ID: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag > message to the phone with > caller_id_name and caller_id_number instead of > effective_caller_id_name and effective_caller_id_number values. > > Thanks, Klaus > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Klaus Hochlehnert > Sent: Sunday, December 06, 2009 4:04 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] caller_id_name + caller_id_number and > Snom 360... > > Hi, > > currently I?m testing the newest FS trunk. > Now I need a hint how to set up an ?old? behavior of version 1.0.4. > > Here?s the scenario: > - Incoming call from caller_id_name: abc and caller_id_number: 123 > - Now I set effective_caller_id_name: xyz and > effective_caller_id_number: 456 > - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) > - After the pickup the Leg B is switching back to the ?old? values > and displays abc + 123 > > But I would rather see the new values during the call (as it was in > version 1.0.4). > What do I need to change? > > Thanks, Klaus > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/73d94204/attachment-0001.html From mailinglist at fribert.dk Sun Dec 6 11:06:45 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 20:06:45 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1C023E020000E1000002A2@mail.fribert.dk> References: <4B1C023E020000E1000002A2@mail.fribert.dk> Message-ID: <4B1C0ED5020000E1000002AC@mail.fribert.dk> Just got the freeswitch started from the command line, and got a bit of logging out I started with the 'multi homed' write up mentioned above (http://wiki.freeswitch.org/wiki/Multi_home_tutorial), as I of course have several nic's on it as it's a firewall. The reason I wanted to use the pfSense firewall, is because I'll get rid of the NAT'in imposed by having it on a local machine. My main reason for setting something up at the first place was to get some of my very limited external IP's back. I've changed the conf/sip_profiles/internal.xml to reflect my LAN addresses My lan is 10.11.12.x My wan is 87.61.18.196 If I start it, it shows a lot on the screen, a line in red is scrolled right out of there, and the buffer isn't large enough to go back 2009-12-06 19:35:42.928556 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 3 0ms 2009-12-06 19:35:42.928564 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 4 0ms 2009-12-06 19:35:42.928573 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 5 0ms 2009-12-06 19:35:42.928583 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 6 0ms 2009-12-06 19:35:42.928592 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 7 0ms 2009-12-06 19:35:42.928600 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 8 0ms 2009-12-06 19:35:42.928609 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 9 0ms 2009-12-06 19:35:42.928619 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 00ms 2009-12-06 19:35:42.928628 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 10ms 2009-12-06 19:35:42.928636 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 20ms 2009-12-06 19:35:42.928922 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_amr] 2009-12-06 19:35:42.928937 [NOTICE] switch_loadable_module.c:182 Adding Codec 'AMR' (AMR) 8000hz 20ms 2009-12-06 19:35:42.930956 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_ilbc] 2009-12-06 19:35:42.930999 [NOTICE] switch_loadable_module.c:182 Adding Codec 'iLBC' (iLBC) 8000hz 30 ms 2009-12-06 19:35:42.931014 [NOTICE] switch_loadable_module.c:182 Adding Codec 'iLBC' (iLBC) 8000hz 20 ms 2009-12-06 19:35:42.938433 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_speex] 2009-12-06 19:35:42.938466 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2009-12-06 19:35:42.938908 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2009-12-06 19:35:42.938931 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2009-12-06 19:35:42.939735 [INFO] mod_siren.c:141 Audio coding: ITU-T Rec. G.722.1, licensed from Pol ycom(R) 2009-12-06 19:35:42.939766 [INFO] mod_siren.c:142 Audio coding: ITU-T Rec. G.722.1 Annex C, licensed from Polycom(R) 2009-12-06 19:35:42.939794 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_siren] 2009-12-06 19:35:42.939810 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 20ms 2009-12-06 19:35:42.939824 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 40ms 2009-12-06 19:35:42.939834 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 60ms 2009-12-06 19:35:42.939843 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 20ms 2009-12-06 19:35:42.939853 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 40ms 2009-12-06 19:35:42.939862 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 60ms 2009-12-06 19:35:42.950262 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_celt] 2009-12-06 19:35:42.950297 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 10ms 2009-12-06 19:35:42.950311 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 8ms 2009-12-06 19:35:42.950322 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 6ms 2009-12-06 19:35:42.950333 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 4ms 2009-12-06 19:35:42.950342 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 2ms 2009-12-06 19:35:42.950351 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 32000hz 10ms 2009-12-06 19:35:42.950804 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_h26x] 2009-12-06 19:35:42.950822 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H264' (H.264 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.950912 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263' (H.263 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.950991 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263-1998' (H.263+ Vid eo (passthru)) 90000hz 0ms 2009-12-06 19:35:42.951072 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263-2000' (H.263++ Vi deo (passthru)) 90000hz 0ms 2009-12-06 19:35:42.951189 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H261' (H.261 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.966090 [INFO] mod_sndfile.c:330 LibSndFile Version : libsndfile-1.0.19 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") WVE (Psion Series 3) (extension "wve") XI (FastTracker 2) (extension "xi") ================================================================================ 2009-12-06 19:35:42.966344 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_sndfile] 2009-12-06 19:35:42.966358 [NOTICE] switch_loadable_module.c:294 Adding File Format 'aiff' 2009-12-06 19:35:42.966500 [NOTICE] switch_loadable_module.c:294 Adding File Format 'au' 2009-12-06 19:35:42.966594 [NOTICE] switch_loadable_module.c:294 Adding File Format 'avr' 2009-12-06 19:35:42.966672 [NOTICE] switch_loadable_module.c:294 Adding File Format 'caf' 2009-12-06 19:35:42.966747 [NOTICE] switch_loadable_module.c:294 Adding File Format 'htk' 2009-12-06 19:35:42.966820 [NOTICE] switch_loadable_module.c:294 Adding File Format 'iff' 2009-12-06 19:35:42.966894 [NOTICE] switch_loadable_module.c:294 Adding File Format 'mat' 2009-12-06 19:35:42.966998 [NOTICE] switch_loadable_module.c:294 Adding File Format 'paf' 2009-12-06 19:35:42.967074 [NOTICE] switch_loadable_module.c:294 Adding File Format 'pvf' 2009-12-06 19:35:42.967148 [NOTICE] switch_loadable_module.c:294 Adding File Format 'raw' 2009-12-06 19:35:42.968115 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sd2' 2009-12-06 19:35:42.968226 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sds' 2009-12-06 19:35:42.968304 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sf' 2009-12-06 19:35:42.968379 [NOTICE] switch_loadable_module.c:294 Adding File Format 'voc' 2009-12-06 19:35:42.968453 [NOTICE] switch_loadable_module.c:294 Adding File Format 'w64' 2009-12-06 19:35:42.968527 [NOTICE] switch_loadable_module.c:294 Adding File Format 'wav' 2009-12-06 19:35:42.968601 [NOTICE] switch_loadable_module.c:294 Adding File Format 'wve' 2009-12-06 19:35:42.968677 [NOTICE] switch_loadable_module.c:294 Adding File Format 'xi' 2009-12-06 19:35:42.968751 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r8' 2009-12-06 19:35:42.968825 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r16' 2009-12-06 19:35:42.968899 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r24' 2009-12-06 19:35:42.968973 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r32' 2009-12-06 19:35:42.969047 [NOTICE] switch_loadable_module.c:294 Adding File Format 'gsm' 2009-12-06 19:35:42.969121 [NOTICE] switch_loadable_module.c:294 Adding File Format 'ul' 2009-12-06 19:35:42.969196 [NOTICE] switch_loadable_module.c:294 Adding File Format 'al' 2009-12-06 19:35:42.969271 [NOTICE] switch_loadable_module.c:294 Adding File Format 'adpcm' 2009-12-06 19:35:42.969751 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_native_fil e] 2009-12-06 19:35:42.969772 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263' 2009-12-06 19:35:42.969886 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G7221' 2009-12-06 19:35:42.969969 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AMR' 2009-12-06 19:35:42.970045 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263-1998' 2009-12-06 19:35:42.970122 [NOTICE] switch_loadable_module.c:294 Adding File Format 'SPEEX' 2009-12-06 19:35:42.970197 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G729' 2009-12-06 19:35:42.970274 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G723' 2009-12-06 19:35:42.970352 [NOTICE] switch_loadable_module.c:294 Adding File Format 'LPC' 2009-12-06 19:35:42.970427 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-16' 2009-12-06 19:35:42.970502 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H261' 2009-12-06 19:35:42.970577 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-16' 2009-12-06 19:35:42.970653 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PCMA' 2009-12-06 19:35:42.970729 [NOTICE] switch_loadable_module.c:294 Adding File Format 'DVI4' 2009-12-06 19:35:42.970804 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-24' 2009-12-06 19:35:42.970880 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PCMU' 2009-12-06 19:35:42.970956 [NOTICE] switch_loadable_module.c:294 Adding File Format 'L16' 2009-12-06 19:35:42.971031 [NOTICE] switch_loadable_module.c:294 Adding File Format 'iLBC' 2009-12-06 19:35:42.971106 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PROXY' 2009-12-06 19:35:42.971217 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PROXY-VID' 2009-12-06 19:35:42.971294 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-24' 2009-12-06 19:35:42.971369 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-32' 2009-12-06 19:35:42.971463 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263-2000' 2009-12-06 19:35:42.971540 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H264' 2009-12-06 19:35:42.971616 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-32' 2009-12-06 19:35:42.971691 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G722' 2009-12-06 19:35:42.971767 [NOTICE] switch_loadable_module.c:294 Adding File Format 'CELT' 2009-12-06 19:35:42.971843 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-40' 2009-12-06 19:35:42.971918 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-40' 2009-12-06 19:35:42.971993 [NOTICE] switch_loadable_module.c:294 Adding File Format 'GSM' 2009-12-06 19:35:42.973983 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_local_stre am] 2009-12-06 19:35:42.974028 [NOTICE] switch_loadable_module.c:270 Adding API Function 'stop_local_stre am' 2009-12-06 19:35:42.975627 [CONSOLE] mod_local_stream.c:142 Can't open directory: /usr/local/freeswit ch/sounds/music/16000 2009-12-06 19:35:42.975972 [CONSOLE] mod_local_stream.c:142 Can't open directory: /usr/local/freeswit ch/sounds/music/32000 2009-12-06 19:35:42.976108 [NOTICE] switch_loadable_module.c:270 Adding API Function 'start_local_str eam' 2009-12-06 19:35:42.976214 [NOTICE] switch_loadable_module.c:270 Adding API Function 'show_local_stre am' 2009-12-06 19:35:42.976297 [NOTICE] switch_loadable_module.c:294 Adding File Format 'local_stream' 2009-12-06 19:35:42.976779 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_tone_strea m] 2009-12-06 19:35:42.976799 [NOTICE] switch_loadable_module.c:294 Adding File Format 'tone_stream' 2009-12-06 19:35:42.976892 [NOTICE] switch_loadable_module.c:294 Adding File Format 'silence_stream' 2009-12-06 19:35:43.96229 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_teletone.so] 2009-12-06 19:35:43.96554 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_core_db.so] 2009-12-06 19:35:43.96828 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_socket.so] 2009-12-06 19:35:43.102288 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_spidermonk ey] 2009-12-06 19:35:43.102327 [NOTICE] switch_loadable_module.c:248 Adding Application 'javascript' 2009-12-06 19:35:43.102515 [NOTICE] switch_loadable_module.c:270 Adding API Function 'jsrun' 2009-12-06 19:35:43.102601 [NOTICE] switch_loadable_module.c:270 Adding API Function 'jsapi' 2009-12-06 19:35:43.110060 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_lua] 2009-12-06 19:35:43.110079 [NOTICE] switch_loadable_module.c:248 Adding Application 'lua' 2009-12-06 19:35:43.110252 [NOTICE] switch_loadable_module.c:270 Adding API Function 'luarun' 2009-12-06 19:35:43.110338 [NOTICE] switch_loadable_module.c:270 Adding API Function 'lua' 2009-12-06 19:35:43.110761 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_say_en] 2009-12-06 19:35:43.110780 [NOTICE] switch_loadable_module.c:395 Adding Say interface 'en' 2009-12-06 19:35:43.110881 [CONSOLE] switch_loadable_module.c:120 Starting runtime thread for CORE_SO FTTIMER_MODULE 2009-12-06 19:35:43.111049 [CONSOLE] switch_loadable_module.c:120 Starting runtime thread for mod_eve nt_socket 2009-12-06 19:35:43.111255 [NOTICE] switch_core.c:898 Created ip list rfc1918.auto default (deny) 2009-12-06 19:35:43.111279 [NOTICE] switch_core.c:906 Created ip list nat.auto default (deny) 2009-12-06 19:35:43.111293 [NOTICE] switch_core.c:914 Created ip list loopback.auto default (deny) 2009-12-06 19:35:43.111348 [NOTICE] switch_core.c:920 Created ip list localnet.auto default (deny) 2009-12-06 19:35:43.111361 [NOTICE] switch_core.c:923 Adding 87.61.18.196/255.255.255.248 (allow) to list localnet.auto 2009-12-06 19:35:43.111380 [CONSOLE] switch_core.c:961 Created ip list lan default (allow) 2009-12-06 19:35:43.111395 [NOTICE] switch_core.c:1064 Adding 10.11.12.0/24 (deny) to list lan 2009-12-06 19:35:43.111422 [NOTICE] switch_core.c:1064 Adding 10.11.12.25/32 (allow) to list lan 2009-12-06 19:35:43.111432 [CONSOLE] switch_core.c:961 Created ip list domains default (deny) 2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 2009-12-06 19:35:43.111986 [CONSOLE] switch_core.c:1465 FreeSWITCH Version 1.0.trunk (13784) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at firewall.fribert.dk> If I then inquire about the internal status I get: freeswitch at firewall.fribert.dk> sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia at 10.11.12.25:5060 BIND-URL sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= If I do a external status it shows: freeswitch at firewall.fribert.dk> sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia at 87.61.18.196:5080 BIND-URL sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= As far as I can see, everything looks ok, except for the 2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 I'm wondering WHY it wants a domain on the external IP??? I then started the SIP softphone, and got: 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute and you must configure your device to use the proper domain in it's authentication credentials. 2009-12-06 19:36:27.988290 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute and you must configure your device to use the proper domain in it's authentication credentials. I've set up the phone to use a domain 10.11.12.25 But I guess, something is screwy with the 'domain' definition as it shows above. So now I'm halted. P.S. I'm full of aw, of the quick and qualified help I've gotten so far, with good pointers to where to start. Thankyou all for your input! This is a brand new line of work for me, so I'm completely at a loss, which is quite fun when I'm used to be the server expert :-D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/d83c0702/attachment-0001.html From freeswitch-users-list at metik.com Sun Dec 6 11:16:43 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 06 Dec 2009 14:16:43 -0500 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> Message-ID: <4B1C031B.8060906@metik.com> Unless the IOS you are running is extremely buggy, "debug voip ccapi" commands should not provide you with that detail, what you really want to use is "debug voip rtp session named-event". Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik > > > You previously stated that your Cisco gateway has some "bug" that > prevents you from using RFC2833, did you enable "dtmf-relay > rtp-nte" on > the voip dial-peer that the call is using? > > > It is a PSTN dialpeer here, and it cannot be defined on it... > > > Unless you have configured the Cisco to support assymetric SDP or are > using a non-default "rtp payload-type nte" setting that does not agree > to well with FS's (default) "rfc2833-pt" setting, you should not > have to > use (SIP) INFO unless you want to. > > I would recommend doing the following to ensure you are hitting the > correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): > > command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) > > > Unfortunately this does not work on PSTN dial peers. > > > > Also, you can sift through "show sip-ua calls" for the call and ensure > that the value of "Negotiated Dtmf-relay" is "rtp-nte". > > > This indeed shows that it has negotiated rtp-nte. Even when I do debug > for CCAPI events (I think) I see it decodes the DTMFs; however, it > ignores them while it accepts them via INFO. As I said: I guess this > is a bug. > > Since the gateway is on a remote site I hesitate on upgrading it until > I hae the chance to go there. > > Thanks, __Yehavi: > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Sun Dec 6 11:29:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:29:14 -0600 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? In-Reply-To: <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> References: <4B1BDA46.7010803@gmx.net> <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> Message-ID: <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> Someone else was asking about this too. I could probably write a dictaction mod in c like the one I made for asterisk starting at about $3k depending on the featureset required. On Dec 6, 2009 10:30 AM, "Peter P GMX" wrote: Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/4e85c5cb/attachment.html From anthony.minessale at gmail.com Sun Dec 6 11:32:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:32:17 -0600 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> References: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> Message-ID: <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> Or set it to true depending on the case Also consider using set_profile_var to set the caller id explicitly instead of using effective. There is also effective_callee_id name and number you could set on the a leg. You'll have to expirement but the one mathieu said is your best bet. On Dec 6, 2009 12:47 PM, "Mathieu Rene" wrote: Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/59976acd/attachment.html From anthony.minessale at gmail.com Sun Dec 6 11:35:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:35:30 -0600 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4B1C031B.8060906@metik.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> <4B1C031B.8060906@metik.com> Message-ID: <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> Some more bad news for you, info dtmf spec has expired and has been abandoned. Wait till you see what they did accept instead...... On Dec 6, 2009 1:22 PM, "Metik" wrote: Unless the IOS you are running is extremely buggy, "debug voip ccapi" commands should not provide you with that detail, what you really want to use is "debug voip rtp session named-event". Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik < freeswitch-users-list at metik.com > > > > You previously stated that your Cisco gateway has some "bug" that > prevents you from us... > ------------------------------------------------------------------------ > > _____________________... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/6965b7f2/attachment.html From yehavi.bourvine at gmail.com Sun Dec 6 11:36:02 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 21:36:02 +0200 Subject: [Freeswitch-users] A few questions about Polycom setup Message-ID: Hello, I have a few questions about Ploycom's usage and provisioning for which I found no answers neither at the docs nor on the WEB: - I would like to enable SIP/TLS. for this I have to import the root certificate. How can I do it via the XML config files? the only method I found is via the phone's interface, but what do you do when you have tens and more of them? - Since the phone is limited to 3way conference I would like it to use a conference room on the server. I've defined: - The result is that when A calls B (the polycom phone) which tries to conference with C is that B does a conference with C and the conference room and A is left on hold... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/c1359a25/attachment.html From anthony.minessale at gmail.com Sun Dec 6 11:41:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:41:38 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> Message-ID: <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> Most of this is unfortunatly because you do not have the proper skill to set it up because, with the proper skills, all of the ways you tried would have ended sucessfully. I say that beacause I have had many users use each of the different methods in your list of failures only they were sucessful. What you are asking for is possible but would require many hours of coding just to help solve your problem. You would have to wait a really long time until someone had the time to do it for free or post a bounty for it. Probably about 1k in consulting time. It may be cheaper for you to pay a consultant to set up one of the ways known to work. These are your options as I see it. On Dec 6, 2009 12:20 PM, "Lon Baker" wrote: Jon, What version of MySQL are you using? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/ddd32fbb/attachment-0001.html From Mailings at kh-dev.de Sun Dec 6 12:37:30 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 21:37:30 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> References: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> Message-ID: Ok, set_profile_var did the trick and also works with intercepted calls. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, December 06, 2009 8:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Or set it to true depending on the case Also consider using set_profile_var to set the caller id explicitly instead of using effective. There is also effective_callee_id name and number you could set on the a leg. You'll have to expirement but the one mathieu said is your best bet. On Dec 6, 2009 12:47 PM, "Mathieu Rene" > wrote: Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/458f5e89/attachment.html From JCasale at activenetwerx.com Sun Dec 6 13:01:23 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 6 Dec 2009 21:01:23 +0000 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1C0ED5020000E1000002AC@mail.fribert.dk> References: <4B1C023E020000E1000002A2@mail.fribert.dk> <4B1C0ED5020000E1000002AC@mail.fribert.dk> Message-ID: >Registrations: >================================================================================================= >================================================================================================= >As far as I can see, everything looks ok, except for the >2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 >I'm wondering WHY it wants a domain on the external IP??? >? >? >I then started the SIP softphone, and got: >? >?2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] >You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute >and you must configure your device to use the proper domain in it's authentication credentials. Yea, it looks like your server is taking the domain of the wan nic. I don't begin to claim I know all there is to know about this (still lurking while I learn as well...) but I got a lab'ed up pfSense box to work only after I edited vars.xml and set: Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something to do with nat and clients in the lan accessing the wan ip. From jbr at consiglia.dk Sun Dec 6 13:33:33 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 6 Dec 2009 22:33:33 +0100 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> Message-ID: The MySQL version is 5.1.37. Well I'm not an expert on every field, and I have no skills in the C, include libraries, and the art of compiling. For this I have to follow the guidelines. But it wouldn't harm the FS project if it generally became more accessible to the race of non-specialists, which I hereby represent. Jon Br?el ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 6. december 2009 20:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db Most of this is unfortunatly because you do not have the proper skill to set it up because, with the proper skills, all of the ways you tried would have ended sucessfully. I say that beacause I have had many users use each of the different methods in your list of failures only they were sucessful. What you are asking for is possible but would require many hours of coding just to help solve your problem. You would have to wait a really long time until someone had the time to do it for free or post a bounty for it. Probably about 1k in consulting time. It may be cheaper for you to pay a consultant to set up one of the ways known to work. These are your options as I see it. On Dec 6, 2009 12:20 PM, "Lon Baker" > wrote: Jon, What version of MySQL are you using? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/bf4d00a1/attachment.html From anthony.minessale at gmail.com Sun Dec 6 14:12:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 16:12:42 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912061407k3b0e6d68p244f25caf1add4bb@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> <191c3a030912061358r79013c9ay445fca25dc24e054@mail.gmail.com> <191c3a030912061400s5994537et178bf2002de06b58@mail.gmail.com> <191c3a030912061405n689a80a5ube645b738db0b531@mail.gmail.com> <191c3a030912061407k3b0e6d68p244f25caf1add4bb@mail.gmail.com> Message-ID: <191c3a030912061412i13349f14n167ffa3dcd873021@mail.gmail.com> Yes, exactly my point. Like I said you have several choices.... be paitent till we have time to code it for free, post a bounty to increase the chance somone will do it from the community, hire someone to set it up for you or keep trying yourself. Did I miss something? On Dec 6, 2009 3:38 PM, "Jon Bruel" wrote: The MySQL version is 5.1.37. Well I?m not an expert on every field, and I have no skills in the C, include libraries, and the art of compiling. For this I have to follow the guidelines. But it wouldn?t harm the FS project if it generally became more accessible to the race of non-specialists, which I hereby represent. *Jon Br?el* ------------------------------ *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 6. december 2009 20:42 *To:* freeswitch-users at lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Lua and database access to core_db Most of this is unfortunatly because you do not have the proper skill to set it up because, wit... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/9be08f45/attachment-0001.html From Prometheus001 at gmx.net Sun Dec 6 14:12:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 23:12:51 +0100 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? In-Reply-To: <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> References: <4B1BDA46.7010803@gmx.net> <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> Message-ID: <4B1C2C63.9080401@gmx.net> Oh that's a lot of money, anybody else needs this feature, so we may share a bounty? Best regards Peter Anthony Minessale schrieb: > > Someone else was asking about this too. > I could probably write a dictaction mod in c like the one I made for > asterisk starting at about $3k depending on the featureset required. > >> On Dec 6, 2009 10:30 AM, "Peter P GMX" > > wrote: >> >> Hello, >> >> I would like to offer a dictation service to a secretary. >> Means: >> >> * the boss is dictating some text on a certain phone number >> * the secretary picks up the recording on the phone and types the >> text into the computer >> >> As the secretary is not able to type in as fastly as heir boss is able >> to speak, she needs some kind of pause and rewind button. >> 1st question: Is there any functionality available for example in >> uuid_broadcast? >> 2nd question: How much would be the effort to implement this >> (uuid_broadcast_pause, uuid_broadcast_UNpause, >> uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. >> >> Best regards >> Peter >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mailinglist at fribert.dk Sun Dec 6 14:53:25 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 23:53:25 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: References: <4B1C023E020000E1000002A2@mail.fribert.dk> <4B1C0ED5020000E1000002AC@mail.fribert.dk> Message-ID: <4B1C43F5020000E1000002B1@mail.fribert.dk> Hi Joseph Ahh, yes, that got rid of that error :-) Now on to the next one. So now it's connecting, both at my provider, and my softphone. Now I have to figure out why it tells me 'Call failed: not found' when I try to call out of the system... But I think that's a task for tomorrow when I'm more awake :-D Thanks! Fribert >>> 06-12-2009 kl. 22:01 skrev "Joseph L. Casale" i meddelelsen : >Registrations: >================================================================================================= >================================================================================================= >As far as I can see, everything looks ok, except for the >2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 >I'm wondering WHY it wants a domain on the external IP??? > > >I then started the SIP softphone, and got: > > 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] >You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute >and you must configure your device to use the proper domain in it's authentication credentials. Yea, it looks like your server is taking the domain of the wan nic. I don't begin to claim I know all there is to know about this (still lurking while I learn as well...) but I got a lab'ed up pfSense box to work only after I edited vars.xml and set: Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something to do with nat and clients in the lan accessing the wan ip. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/b9ad9246/attachment.html From sklein at singular.com Sat Dec 5 19:21:16 2009 From: sklein at singular.com (Steve Klein) Date: Sat, 5 Dec 2009 19:21:16 -0800 Subject: [Freeswitch-users] lua+sqlite example? Message-ID: <04a201ca7623$2c0b2020$84216060$@com> Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/51e80e09/attachment.html From sklein at singular.com Sun Dec 6 12:07:39 2009 From: sklein at singular.com (Steve Klein) Date: Sun, 6 Dec 2009 12:07:39 -0800 Subject: [Freeswitch-users] Database suggestions/pointers/? Message-ID: <053d01ca76af$c311d2c0$49357840$@com> Greetings. We need to add database access to an IVR application we are prototyping. Based on FS "best practice" suggestions, we are using Lua for the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite would be a recommended approach. However, we can't find any examples of this combo anywhere. So, what is the "best practice" scripting + database recommendation for a high-volume database-driven FS app? Any help/pointers greatly appreciated! --Steve Klein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/3261165c/attachment.html From timuckun at gmail.com Sun Dec 6 16:59:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 7 Dec 2009 13:59:31 +1300 Subject: [Freeswitch-users] Database suggestions/pointers/? In-Reply-To: <053d01ca76af$c311d2c0$49357840$@com> References: <053d01ca76af$c311d2c0$49357840$@com> Message-ID: <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein wrote: > Greetings. We need to add database access to an IVR application we are > prototyping. Based on FS ?best practice? suggestions, we are using Lua for > the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite > would be a recommended approach. However, we can?t find any examples of this > combo anywhere. So, what is the ?best practice? scripting + database > recommendation for a high-volume database-driven FS app? > I would suggest you take a look at freeswitcher (http://github.com/bougyman/freeswitcher). The good thing is that it's ruby and therefore you can use any database compatible with ruby (that's all of them pretty much). You can also use an ORM of your choice or if you don't want to use an ORM you can use the amazingly fantastic sequel library. Being ruby it will run outside of the freeswitch memory space and you will have to use the inbound/outbound socket API. That may be a good thing if you want to separate your database and IVR logic from the machine running your freeswitch. Ruby is pretty easy to pick up if you don't know it and there are a wealth of libraries if you want to do other things like connect to web sites, manipulate XML, etc. There is also a liverpie http://github.com/jsgoecke/liverpie which is more of a proxy thing you can interface with any language. I am sure lua is nice but it seems like people are having some problems with ODBC, memory leaks etc when it comes to databases. If you go a ruby library that all goes away. From djbinter at yahoo.com Sun Dec 6 17:17:14 2009 From: djbinter at yahoo.com (DJB) Date: Sun, 6 Dec 2009 17:17:14 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. Message-ID: <168319.49226.qm@web37502.mail.mud.yahoo.com> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -------------------------------------------------------------------------------------------------------------- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? S References: <41A44DD027064988A914974405788C2E@procent> <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com> <1254901192035-3780245.post@n2.nabble.com> <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com> <1255008427639-3788019.post@n2.nabble.com> <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com> <1255169044209-3799274.post@n2.nabble.com> <1255337256919-3806786.post@n2.nabble.com> <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org> <1255363492193-3808860.post@n2.nabble.com> Message-ID: This bug has been now closed out in jira due to no response for requested information. If you wish to resolve this issue please follow up on your bugs when information is requested. Mike On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote: > > Nope, I wanted to make sure that this is indeed a bug. I opened an issue in > JIRA before regarding some other matter and it turned out to be my mistake, > so I decided to try mailing list first this time. > MA > > > > Brian West wrote: >> >> Did you open a jira and attach all the info? >> >> /b >> >> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote: >> >>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src == >>> ). RTP packets are sent every 20ms. >>> >>> MAniserowicz From jaybinks at gmail.com Sun Dec 6 20:04:03 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 7 Dec 2009 14:04:03 +1000 Subject: [Freeswitch-users] Audiocodes PRI Gateway Message-ID: Guys, im after info from people with experience with AudioCodes Mediant 2k PRI Gateways. specifically how well they inter-op with Freeswitch, and how compliant their SIP stack is. I guess the bottom line is, would you recommend these gateways or would you suggest something else ? -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/92522e65/attachment.html From dujinfang at gmail.com Sun Dec 6 20:45:54 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Dec 2009 12:45:54 +0800 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario Message-ID: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 Thanks. From imthiyazg at gmail.com Sun Dec 6 21:52:46 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Mon, 7 Dec 2009 11:22:46 +0530 Subject: [Freeswitch-users] Audiocodes PRI Gateway In-Reply-To: References: Message-ID: <8595daf70912062152v122d3acdvcd84db8162384ee4@mail.gmail.com> We are using Audiocodes and Sangoma netborder express GW with Freeswitch . it works well. Thanks Imthiyaz On Mon, Dec 7, 2009 at 9:34 AM, jay binks wrote: > Guys, > ??im after info from people with experience with AudioCodes Mediant 2k PRI > Gateways. > specifically?how well they?inter-op?with Freeswitch, and how compliant their > SIP stack is. > I guess the bottom line is, would you?recommend?these gateways or would you > suggest something else ? > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From mike at jerris.com Sun Dec 6 21:53:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 7 Dec 2009 00:53:33 -0500 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario In-Reply-To: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> References: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> Message-ID: <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> Please report bugs to jira.freeswitch.org. Mike On Dec 6, 2009, at 11:45 PM, Seven Du wrote: > Hi, > > I know there's some chang on att_xfer, and after upgrade(re-bootstrap) > to trunk code, no sound after att_xfer. > > Then I rebuild FS 15807 with a fresh checkout, but still using the old > conf/ settings, sound is ok, but there are other problems: > > A call B, and B att_xfer C > > 1) origination_cancel_key not working. no even no DTMF log in FS when > I press # or any other key, I tried with Zoiper and Snom(on the B leg) > 2) when C answers, B immediately hangup, so B has no chance talk to C > > Could this be a problem? I pasted logs: > > http://pastebin.freeswitch.org/11417 From abeka at greatiam.com Sun Dec 6 23:30:45 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Mon, 07 Dec 2009 07:30:45 +0000 Subject: [Freeswitch-users] Mutual Registration of servers Message-ID: <4B1CAF25.6010706@greatiam.com> Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. From dujinfang at gmail.com Mon Dec 7 02:21:29 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Dec 2009 18:21:29 +0800 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario In-Reply-To: <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> References: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> Message-ID: <23f91030912070221v3be54ed5x2a1ac1e6c9bf0e7d@mail.gmail.com> Thanks, done. 2009/12/7 Michael Jerris : > Please report bugs to jira.freeswitch.org. > > Mike > > On Dec 6, 2009, at 11:45 PM, Seven Du wrote: > >> Hi, >> >> I know there's some chang on att_xfer, and after upgrade(re-bootstrap) >> to trunk code, no sound after att_xfer. >> >> Then I rebuild FS 15807 with a fresh checkout, but still using the old >> conf/ settings, sound is ok, but there are other problems: >> >> A call B, and B att_xfer C >> >> 1) origination_cancel_key not working. no even no DTMF log in FS when >> I press # or any other key, I tried with Zoiper and Snom(on the B leg) >> 2) when C answers, B immediately hangup, so B has no chance talk to C >> >> Could this be a problem? I pasted logs: >> >> http://pastebin.freeswitch.org/11417 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lakindia89 at gmail.com Mon Dec 7 05:15:10 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 7 Dec 2009 18:45:10 +0530 Subject: [Freeswitch-users] Need Conference design help In-Reply-To: <26653473.post@talk.nabble.com> References: <26653473.post@talk.nabble.com> Message-ID: <7d79b3930912070515p8a5aae3tc8dc51f44dd6d80e@mail.gmail.com> Have a look at mod_conference http://wiki.freeswitch.org/wiki/Mod_conference On Sat, Dec 5, 2009 at 12:47 PM, shehzad p wrote: > > Hello Every one, > > I have to design conference, and I need community guidance to efficiently > accomplish that. > > I need to create Conference which will have three kind of users: > 1. Moderator (may be only one per conference) > 2. User who can participate in conference without moderator interaction. > 3. User who can only participate when Moderator allow them to get in. > > Also besides above setup I have to perform other things like Record the > conference, Multicast the conference to other freeswitch server. I saw the > conference Record CLI command but wondering where to setup when conference > starts. I am also wondering how Multicast Conference is possible in > Freeswitch and how the receiver Freeswitch configuration will look like. > > Thanks. > msp > > -- > View this message in context: > http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ba31c985/attachment.html From erandr-junk at usa.net Mon Dec 7 07:28:52 2009 From: erandr-junk at usa.net (eaf) Date: Mon, 7 Dec 2009 07:28:52 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> Message-ID: <26678873.post@talk.nabble.com> Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently sitting in the first line of the top chart showing 3% CPU utilization when idle. Don't know whether it's because of the remaining epolls in sofia or whether it's because there are still some threads left in freeswitch that I neglected to change because they were sleeping with 100ms interval, so I figured, who cares. Maybe when all things come together (sofia, 100ms*N) freeswitch ends up spending 3% of CPU while doing pretty much nothing. Btw, compared with Asterisk, the latter is not even visible on the first top's screen and spends 1% CPU when bridging two G711 calls and recording them to disk. So, at this time I have both original Asterisk and FS setups running. One is seemless but clumsy in configuration, the other one is neat and stylish but too preoccupied with smth... Should I look into sofia epollers? That's kind of deep in the code. Or should I just stick with Asterisk? Anthony Minessale-2 wrote: > > There is another user here with a 300mhz box. I am willing to investigate > this improved performance for weak devices but I need to do it in a sane > cross-platform way. > > > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > > wrote: > >> A word to the wise to the general FreeSWITCH community: If Anthony >> Minessale suggests that you try to do any number of things, it's a very >> good idea to try all those ideas before continuing on. I've known him, >> MikeJ, and bkw for several years, and they almost always have very good >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely >> frustrating to try to help people out who won't try the provided >> suggestions first. >> >> And note directly to "eaf" - bogomips is quite possibly the least >> significant bit of data about a cpu that you will get out of >> /proc/cpuinfo... The name itself - bogo, means bogus. >> http://en.wikipedia.org/wiki/Bogomips >> >> -Yossi >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Mon Dec 7 07:44:24 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 7 Dec 2009 07:44:24 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <2FE67B50CF5C456E9958B83513618E3F@greyhawk.tonecommander.com> I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/08a565af/attachment.html From anthony.minessale at gmail.com Mon Dec 7 08:00:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 10:00:17 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26678873.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> Message-ID: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > Here is what I found... > > I tried high-priority scheduling as per your suggestion, reniced the > program > explicitly, rewrote timer thread to sleep on cond. variable and activate > only when there are timers and only when the timer actually had to be > clicked, turned off SQL thread and removed polling from sofia profile > thread. > > That pretty much eliminated all idle 1ms sleepers that were there except > for > three in sofia itself (su_epoll_port). And when I was about to be happy, I > found that two outgoing calls through my VOIP providers when bridged > together showed terrible distortions. I undid all my changes, tried 1.0.4, > trunk (noticed btw that when I bridge two calls via loopback in JS in the > trunk I must keep JS running, or the calls get terminated - NOT the same as > in 1.0.4 where exitting JS left calls running), got pretty much the same > sad > results. At the same time calls bridged by freeswitch between LAN and any > of > the VOIP providers behaved just fine. And calls bridged by Asterisk any way > were fine too. So that pretty much looked like the end of the freeswitch > trials for me. > > But then I timed your code, mine and found that all those 1ms sleeps that > your timer thread was doing (and all those pollers were doing as well) were > actually 4ms sleeps because you know what unless kernel is configured with > HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > (HZ=100). Mine was 250. > > This actually meant that the original timer thread was firing once, > sleeping > for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 > times back-to-back, etc. It was still firing 20ms timers on time, but 30ms > ones of course were not, since 30ms doesn't divide by 4 evenly. Plus > whoever > relied on runtime.reference or switch_micro_time_now() were kind of screwed > because both were running jumpy. Plus whoever assumed that apr_sleep(1000) > or cond_yield() was sleeping for 1ms were also in for a surprise. It felt > satisfying to find that, however it didn't explain why the same distortions > were observed with rewritten timer thread and disabled RTP timers. > > Anyway, I sighed (pretty much like you) and recompiled the kernel with > HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes > south, > you need to hook up serial console and see what the heck went wrong. > > That eliminated distortions, ha! But made freeswitch more CPU hungry. Now > the remaining 1ms threads sitting in sofia epoll were really polling for > 1ms, not 4, and freeswitch was consistently sitting in the first line of > the > top chart showing 3% CPU utilization when idle. > > Don't know whether it's because of the remaining epolls in sofia or whether > it's because there are still some threads left in freeswitch that I > neglected to change because they were sleeping with 100ms interval, so I > figured, who cares. Maybe when all things come together (sofia, 100ms*N) > freeswitch ends up spending 3% of CPU while doing pretty much nothing. > > Btw, compared with Asterisk, the latter is not even visible on the first > top's screen and spends 1% CPU when bridging two G711 calls and recording > them to disk. > > So, at this time I have both original Asterisk and FS setups running. One > is > seemless but clumsy in configuration, the other one is neat and stylish but > too preoccupied with smth... Should I look into sofia epollers? That's kind > of deep in the code. Or should I just stick with Asterisk? > > > > > > Anthony Minessale-2 wrote: > > > > There is another user here with a 300mhz box. I am willing to > investigate > > this improved performance for weak devices but I need to do it in a sane > > cross-platform way. > > > > > > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > > >> wrote: > > > >> A word to the wise to the general FreeSWITCH community: If Anthony > >> Minessale suggests that you try to do any number of things, it's a very > >> good idea to try all those ideas before continuing on. I've known him, > >> MikeJ, and bkw for several years, and they almost always have very good > >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > >> frustrating to try to help people out who won't try the provided > >> suggestions first. > >> > >> And note directly to "eaf" - bogomips is quite possibly the least > >> significant bit of data about a cpu that you will get out of > >> /proc/cpuinfo... The name itself - bogo, means bogus. > >> http://en.wikipedia.org/wiki/Bogomips > >> > >> -Yossi > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/a83d6006/attachment-0001.html From mike at jerris.com Mon Dec 7 08:16:30 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 7 Dec 2009 11:16:30 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: Also I have seen some people reporting that the new tickless timers in newer kernels work better. You may want to try those. Mike On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others work better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > Here is what I found... > > I tried high-priority scheduling as per your suggestion, reniced the program > explicitly, rewrote timer thread to sleep on cond. variable and activate > only when there are timers and only when the timer actually had to be > clicked, turned off SQL thread and removed polling from sofia profile > thread. > > That pretty much eliminated all idle 1ms sleepers that were there except for > three in sofia itself (su_epoll_port). And when I was about to be happy, I > found that two outgoing calls through my VOIP providers when bridged > together showed terrible distortions. I undid all my changes, tried 1.0.4, > trunk (noticed btw that when I bridge two calls via loopback in JS in the > trunk I must keep JS running, or the calls get terminated - NOT the same as > in 1.0.4 where exitting JS left calls running), got pretty much the same sad > results. At the same time calls bridged by freeswitch between LAN and any of > the VOIP providers behaved just fine. And calls bridged by Asterisk any way > were fine too. So that pretty much looked like the end of the freeswitch > trials for me. > > But then I timed your code, mine and found that all those 1ms sleeps that > your timer thread was doing (and all those pollers were doing as well) were > actually 4ms sleeps because you know what unless kernel is configured with > HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > (HZ=100). Mine was 250. > > This actually meant that the original timer thread was firing once, sleeping > for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 > times back-to-back, etc. It was still firing 20ms timers on time, but 30ms > ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever > relied on runtime.reference or switch_micro_time_now() were kind of screwed > because both were running jumpy. Plus whoever assumed that apr_sleep(1000) > or cond_yield() was sleeping for 1ms were also in for a surprise. It felt > satisfying to find that, however it didn't explain why the same distortions > were observed with rewritten timer thread and disabled RTP timers. > > Anyway, I sighed (pretty much like you) and recompiled the kernel with > HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, > you need to hook up serial console and see what the heck went wrong. > > That eliminated distortions, ha! But made freeswitch more CPU hungry. Now > the remaining 1ms threads sitting in sofia epoll were really polling for > 1ms, not 4, and freeswitch was consistently sitting in the first line of the > top chart showing 3% CPU utilization when idle. > > Don't know whether it's because of the remaining epolls in sofia or whether > it's because there are still some threads left in freeswitch that I > neglected to change because they were sleeping with 100ms interval, so I > figured, who cares. Maybe when all things come together (sofia, 100ms*N) > freeswitch ends up spending 3% of CPU while doing pretty much nothing. > > Btw, compared with Asterisk, the latter is not even visible on the first > top's screen and spends 1% CPU when bridging two G711 calls and recording > them to disk. > > So, at this time I have both original Asterisk and FS setups running. One is > seemless but clumsy in configuration, the other one is neat and stylish but > too preoccupied with smth... Should I look into sofia epollers? That's kind > of deep in the code. Or should I just stick with Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/31a8a586/attachment.html From lfurrea at gmail.com Mon Dec 7 08:28:38 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Mon, 7 Dec 2009 10:28:38 -0600 Subject: [Freeswitch-users] Sporadic call drops In-Reply-To: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> References: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> Message-ID: I will certainly shchedule time for the upgrade. Thanks for the answer On Fri, Dec 4, 2009 at 1:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > we changed that message a long time ago so people would not think that > anymore > We are now 3000 rev beyond the version you are at, I would like it if you > try the lastest trunk. > > > On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea wrote: > >> Hi all, >> >> Guys I know the question could be too vague, but I have a customer that >> just reported frequent failure to place outbound calls though a PSTN gateway >> on the LAN. >> >> I looked at the logs and I seem to be able to confirm that FS fails to >> place the call through the gateway and that the issue resides on the FS side >> since the first channel that s killed is tht of the internal extension >> registered to FS and then FS send the BYE to gw and kills the channel. >> >> What are possible causes of this? >> >> I know you always like to look at complete logs but here's a snip that >> could shed some light on the disconnection. (I can provide full logs if >> required and worthed) >> >> 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() >> Channel sofia/internal/200 at 172.16.3.5 entering state [ready][200] >> 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() >> Channel sofia/internal/200 at 172.16.3.5 entering state [terminated][200] >> 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() >> Hangup sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] >> 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal sofia/internal/ >> 200 at 172.16.3.5 [BREAK] >> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() >> sofia/internal/200 at 172.16.3.5 ending bridge by request from write >> function >> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() >> sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] >> >> >> Is the 6th line normal behavior for ending the channel? >> >> FreeSWITCH Version 1.0.trunk (13484M) >> >> TIA >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4db8a923/attachment.html From djbinter at yahoo.com Mon Dec 7 08:42:57 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 08:42:57 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <168319.49226.qm@web37502.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> Message-ID: <987536.45831.qm@web37508.mail.mud.yahoo.com> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. ________________________________ From: DJB To: FREESWITCH-USERS MAILING LIST Sent: Sun, December 6, 2009 5:17:14 PM Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -------------------------------------------------------------------------------------------------------------- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? S References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > Any suggestion would be appreciated. Thank you. > > ------------------------------ > *From:* DJB > *To:* FREESWITCH-USERS MAILING LIST > > *Sent:* Sun, December 6, 2009 5:17:14 PM > *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with > high priority enabled. > > I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version > 1.0.4 (exported) with only one thing difference which is the first one is > running with -hp enabled; however, I have noticed that the one with -hp > option consumed double in memory usage than the other one. > > I wonder whether anyone can explain why. Thank you. > > Please see below: > > > -------------------------------------------------------------------------------------------------------------- > top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 > Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie > Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers > Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch > > 4418 session(s) 14/100 > > root 30750 2.1 *19.9* 1879252 1634300 ? S /usr/local/freeswitch/bin/freeswitch -nc -hp > > > -------------------------------------------------------------------------------------------------------------- > top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 > Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie > Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers > Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch > > 4478 session(s) 14/100 > > root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 > /usr/local/freeswitch/bin/freeswitch -nc > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/66fc8548/attachment.html From djbinter at yahoo.com Mon Dec 7 09:12:18 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 09:12:18 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> Message-ID: <794775.77398.qm@web37501.mail.mud.yahoo.com> Anthony, Thank you for your clear response. Based on your recommendation, if I want to route more calls to the first server, should I take off "-hp", or it's better to run with it. We are running FS for pass-thru traffic with signaling only. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Mon, December 7, 2009 8:56:14 AM Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. > > > ________________________________ From: DJB >To: FREESWITCH-USERS MAILING LIST >Sent: Sun, December 6, 2009 5:17:14 PM >Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. > > > >I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. > > >I wonder whether anyone can explain why. Thank you. > > >Please see below: > > >-------------------------------------------------------------------------------------------------------------- >top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 >Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, > 0 zombie >Cpu(s): 0.9%us, 0.2%sy, > 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st >Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers >Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch > > >4418 session(s) 14/100 > > >root 30750 2.1 19.9 1879252 1634300 > ? S > >-------------------------------------------------------------------------------------------------------------- >>top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 >Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st >Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers >Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch > > >4478 session(s) 14/100 > > >root 7147 1.9 9.4 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc > > > > > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/46f0e3dc/attachment.html From anthony.minessale at gmail.com Mon Dec 7 09:31:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:31:27 -0600 Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <794775.77398.qm@web37501.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> <794775.77398.qm@web37501.mail.mud.yahoo.com> Message-ID: <191c3a030912070931j5c96576fsede73fd56bedeca0@mail.gmail.com> maybe you can try both ways and see if there is a significant difference? I think -hp would help more if you were doing media than if you were not but that does not mean it could not still help performance but really the extra performance would only show up once you had consumed all the resources the box had to offer without -hp enabled in most cases. On Mon, Dec 7, 2009 at 11:12 AM, DJB wrote: > Anthony, > > Thank you for your clear response. Based on your recommendation, if I want > to route more calls to the first server, should I take off "-hp", or it's > better to run with it. We are running FS for pass-thru traffic with > signaling only. > > > ------------------------------ > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Mon, December 7, 2009 8:56:14 AM > *Subject:* Re: [Freeswitch-users] Question regarding running FreeSWITCH > with high priority enabled. > > One of the properties of -hp is to enable memlockall() which means disable > swapping. This causes all memory used by FS to be resident permanently and > is much more costly in memory usage. -hp also uses a RR scheduler runs the > process at a less nice level and increases a few other process ulimits. > This mode is designed for high end usage and uses more resources when idle > with a large payout when scaling to many calls. > > > > > On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: > >> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting >> calls with load balancing from another switch. Thus, the traffic type are >> pretty much identical and both FSs have exactly the same on configuration. >> Any suggestion would be appreciated. Thank you. >> >> ------------------------------ >> *From:* DJB >> *To:* FREESWITCH-USERS MAILING LIST < >> freeswitch-users at lists.freeswitch.org> >> *Sent:* Sun, December 6, 2009 5:17:14 PM >> *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with >> high priority enabled. >> >> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version >> 1.0.4 (exported) with only one thing difference which is the first one is >> running with -hp enabled; however, I have noticed that the one with -hp >> option consumed double in memory usage than the other one. >> >> I wonder whether anyone can explain why. Thank you. >> >> Please see below: >> >> >> -------------------------------------------------------------------------------------------------------------- >> top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 >> Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >> Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, >> 0.0%st >> Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers >> Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch >> >> 4418 session(s) 14/100 >> >> root 30750 2.1 *19.9* 1879252 1634300 ? S> /usr/local/freeswitch/bin/freeswitch -nc -hp >> >> >> -------------------------------------------------------------------------------------------------------------- >> top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 >> Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >> Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, >> 0.0%st >> Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers >> Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch >> >> >> 4478 session(s) 14/100 >> >> root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 >> /usr/local/freeswitch/bin/freeswitch -nc >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/e05eca83/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 7 09:32:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:32:25 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: References: <26594250.post@talk.nabble.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <191c3a030912070932m1c17b98dnd09181dcfe53d1f1@mail.gmail.com> oh and also use top -H to see which threads are using specific CPU and try to cross reference them by attaching with gdb and dumping all the thread bt On Mon, Dec 7, 2009 at 10:16 AM, Michael Jerris wrote: > Also I have seen some people reporting that the new tickless timers in > newer kernels work better. You may want to try those. > > Mike > > On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: > > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm. We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using. I am curious > what would happen if you install Kristian's astlinux on one of your devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do > these systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread > which will make all the switch_cond_next share a 1ms conditional instead of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others > work better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > >> >> Here is what I found... >> >> I tried high-priority scheduling as per your suggestion, reniced the >> program >> explicitly, rewrote timer thread to sleep on cond. variable and activate >> only when there are timers and only when the timer actually had to be >> clicked, turned off SQL thread and removed polling from sofia profile >> thread. >> >> That pretty much eliminated all idle 1ms sleepers that were there except >> for >> three in sofia itself (su_epoll_port). And when I was about to be happy, I >> found that two outgoing calls through my VOIP providers when bridged >> together showed terrible distortions. I undid all my changes, tried 1.0.4, >> trunk (noticed btw that when I bridge two calls via loopback in JS in the >> trunk I must keep JS running, or the calls get terminated - NOT the same >> as >> in 1.0.4 where exitting JS left calls running), got pretty much the same >> sad >> results. At the same time calls bridged by freeswitch between LAN and any >> of >> the VOIP providers behaved just fine. And calls bridged by Asterisk any >> way >> were fine too. So that pretty much looked like the end of the freeswitch >> trials for me. >> >> But then I timed your code, mine and found that all those 1ms sleeps that >> your timer thread was doing (and all those pollers were doing as well) >> were >> actually 4ms sleeps because you know what unless kernel is configured with >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms >> (HZ=100). Mine was 250. >> >> This actually meant that the original timer thread was firing once, >> sleeping >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 >> times back-to-back, etc. It was still firing 20ms timers on time, but 30ms >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus >> whoever >> relied on runtime.reference or switch_micro_time_now() were kind of >> screwed >> because both were running jumpy. Plus whoever assumed that apr_sleep(1000) >> or cond_yield() was sleeping for 1ms were also in for a surprise. It felt >> satisfying to find that, however it didn't explain why the same >> distortions >> were observed with rewritten timer thread and disabled RTP timers. >> >> Anyway, I sighed (pretty much like you) and recompiled the kernel with >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes >> south, >> you need to hook up serial console and see what the heck went wrong. >> >> That eliminated distortions, ha! But made freeswitch more CPU hungry. Now >> the remaining 1ms threads sitting in sofia epoll were really polling for >> 1ms, not 4, and freeswitch was consistently sitting in the first line of >> the >> top chart showing 3% CPU utilization when idle. >> >> Don't know whether it's because of the remaining epolls in sofia or >> whether >> it's because there are still some threads left in freeswitch that I >> neglected to change because they were sleeping with 100ms interval, so I >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. >> >> Btw, compared with Asterisk, the latter is not even visible on the first >> top's screen and spends 1% CPU when bridging two G711 calls and recording >> them to disk. >> >> So, at this time I have both original Asterisk and FS setups running. One >> is >> seemless but clumsy in configuration, the other one is neat and stylish >> but >> too preoccupied with smth... Should I look into sofia epollers? That's >> kind >> of deep in the code. Or should I just stick with Asterisk? >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/89804735/attachment.html From anthony.minessale at gmail.com Mon Dec 7 09:35:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:35:04 -0600 Subject: [Freeswitch-users] lua+sqlite example? In-Reply-To: <04a201ca7623$2c0b2020$84216060$@com> References: <04a201ca7623$2c0b2020$84216060$@com> Message-ID: <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein wrote: > Greetings. We are attempting to add sqlite access to an IVR application > we are prototyping. We are using lua for the scripts. Is there an example > anywhere of a lua + sqlite script? Do we need to install luasql? Any > help/pointers greatly appreciated. > > > > --Steve Klein > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d5be5e76/attachment.html From msc at freeswitch.org Mon Dec 7 09:43:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 09:43:17 -0800 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <4B1CAF25.6010706@greatiam.com> References: <4B1CAF25.6010706@greatiam.com> Message-ID: <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah wrote: > Pardon me if this has been addressed already. > How does one go about having in the simplest instance 2 servers > registering with each other on startup whereby the users registering > would be able to call each other. > The 2 servers are in different domains. > > Thanks. > Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/c2605e5c/attachment.html From ken at ksac.com Mon Dec 7 09:27:49 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 09:27:49 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 Message-ID: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/2db542cd/attachment-0001.html From brian at freeswitch.org Mon Dec 7 10:10:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 12:10:11 -0600 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get > esl to compile. My version is earlier than the snow leopard that is > mentioned in the general install docs, and I have tried it with and > without the compiler flags in the freewswtch installation -> MAC os X. > I have also googled this, and don?t see what I am doing wrong. > Anybody there that can help? > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/ > freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../ > libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable - > Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="- > I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g - > ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc - > lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > _main > __convert_to_string > __efree > __emalloc > __estrndup > __zend_get_parameters_array_ex > __zend_list_find > __zval_copy_ctor > _compiler_globals > _convert_to_long > _zend_error > _zend_get_constant > _zend_hash_find > _zend_register_list_destructors_ex > _zend_register_long_constant > _zend_register_resource > _zend_rsrc_list_get_rsrc_type > _zend_wrong_param_count > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make: *** [phpmod] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d9a2f13a/attachment.html From mcampbellsmith at gmail.com Mon Dec 7 10:11:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 05:11:17 +1100 Subject: [Freeswitch-users] Access to users variables Message-ID: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> Hi! How can I access the variables that are defined in a users xml file? For example, say user 1000 has a variable called smsnumber, as defined below: How can I use variable ${smsnumber} in a dialplan to run a perl script using ? Thanks From msc at freeswitch.org Mon Dec 7 10:21:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:21:36 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> Forgive me if I ask the obvious questions... Did you "make" in src/libs/esl before doing "make phpmod" ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer wrote: > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get esl to > compile. My version is earlier than the snow leopard that is mentioned in > the general install docs, and I have tried it with and without the compiler > flags in the freewswtch installation -> MAC os X. > > I have also googled this, and don?t see what I am doing wrong. Anybody > there that can help? > > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" > CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE > -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc > -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > > _main > > __convert_to_string > > __efree > > __emalloc > > __estrndup > > __zend_get_parameters_array_ex > > __zend_list_find > > __zval_copy_ctor > > _compiler_globals > > _convert_to_long > > _zend_error > > _zend_get_constant > > _zend_hash_find > > _zend_register_list_destructors_ex > > _zend_register_long_constant > > _zend_register_resource > > _zend_rsrc_list_get_rsrc_type > > _zend_wrong_param_count > > collect2: ld returned 1 exit status > > make[1]: *** [ESL.so] Error 1 > > make: *** [phpmod] Error 2 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/6825e5d6/attachment.html From msc at freeswitch.org Mon Dec 7 10:25:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:25:36 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> Message-ID: <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > How can I access the variables that are defined in a users xml file? > > For example, say user 1000 has a variable called smsnumber, as defined > below: > > > > > > > > > > > > > How can I use variable ${smsnumber} in a dialplan to run a perl script > using ? > > Do you just want to pass the value in smsnumber to the sms.pl script? Have you tried this? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/b82afa81/attachment.html From Prometheus001 at gmx.net Mon Dec 7 10:31:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 07 Dec 2009 19:31:36 +0100 Subject: [Freeswitch-users] continue_on_fail Message-ID: <4B1D4A08.80507@gmx.net> I have a Problem with continue_on_fail. I have setup a hunt group and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter From ken at ksac.com Mon Dec 7 10:33:44 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 10:33:44 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> References: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> Message-ID: I did make first, but did not install any extra dev stuff, thinking I already had them. Is there a way to turn on verbose and finding out exactly what it no there that is expected? Thanksmuch!! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, December 07, 2009 1:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 Forgive me if I ask the obvious questions... Did you "make" in src/libs/esl before doing "make phpmod" ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer > wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d476470b/attachment-0001.html From ken at ksac.com Mon Dec 7 10:34:41 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 10:34:41 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, December 07, 2009 1:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/e0470b04/attachment.html From mcampbellsmith at gmail.com Mon Dec 7 10:37:03 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 05:37:03 +1100 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> Message-ID: <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> Hi! That's exactly what I want to do and that was the first thing I tried, but nothing is passed to the script. In a case like this, what defines if variable smsnumber is taken from the A path or B path? (The A path does not have smsnumber defined) On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins wrote: > > > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> How can I access the variables that are defined in a users xml file? >> >> For example, say user 1000 has a variable called smsnumber, as defined >> below: >> >> >> ? >> ? ? >> ? ? ? >> ? ? >> ? ? >> ? ? ? >> ? ? >> ? >> >> >> How can I use variable ${smsnumber} in a dialplan to run a perl script >> using ? >> > > Do you just want to pass the value in smsnumber to the sms.pl script? Have > you tried this? > > > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jmesquita at freeswitch.org Mon Dec 7 10:44:34 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 7 Dec 2009 16:44:34 -0200 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> Message-ID: Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > I installed libjpeg-7 following this website: > http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And > the previous error is replaced by a new one: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. > -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF > .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o > at_interpreter.o > at_interpreter.c: In function ???command_search???: > at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use > in this function) > at_interpreter.c:5299: error: (Each undeclared identifier is reported only > once > at_interpreter.c:5299: error: for each function it appears in.) > at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in > this function) > at_interpreter.c: In function ???at_interpreter???: > at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in > this function) > make[8]: *** [at_interpreter.lo] Error 1 > > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > However, I'm still able to start freeswitch and mod_skypiax and make skype > calls with no problem. > > Regards, > -Jingwei > > > > On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > >> No, I didn't change or update the system libs. I just wanted to double >> check whether my system has this libjpeg library. ./configure was definitely >> executed before the source codes were rebuilt. >> >> Regards, >> -Jingwei >> >> >> On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: >> >>> Hi, >>> >>> That one is on your side. If you changed/updated system libs it might be >>> worth doing another ./configure >>> >>> Cheers, >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >>> >>> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >>> However, I encounter another one. >>> >>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>> -lc >>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: >>> cannot open shared object file: No such file or directory >>> make[8]: *** [at_interpreter_dictionary.h] Error 127 >>> make[7]: *** [all] Error 2 >>> make[6]: *** [all-recursive] Error 1 >>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_voipcodecs-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> Do you have idea about this one? >>> >>> Thanks! >>> >>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >>> >>>> Consider it fixed. >>>> Committed revision 15765. >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>>> >>>> Hi Guys, >>>> >>>> I got a compilation error of skypiax_protocol.c with the latest version >>>> r15764. >>>> >>>> Compiling skypiax_protocol.c... >>>> *cc1: warnings being treated as errors* >>>> skypiax_protocol.c: In function ???X11_errors_handler???: >>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c: In function ???skypiax_send_message???: >>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >>>> code >>>> make[5]: *** [skypiax_protocol.o] Error 1 >>>> make[4]: *** [install] Error 1 >>>> make[3]: *** [mod_skypiax-install] Error 1 >>>> make[2]: *** [install-recursive] Error 1 >>>> >>>> I personally checked the file and it shouldn't be a merge problem. Does >>>> anyone encounter this as well? >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4183f10a/attachment-0001.html From jerry.richards at teotech.com Mon Dec 7 10:49:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 7 Dec 2009 10:49:01 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/73e6eaec/attachment.html From abeka at greatiam.com Mon Dec 7 10:50:58 2009 From: abeka at greatiam.com (Otis) Date: Mon, 07 Dec 2009 18:50:58 +0000 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> References: <4B1CAF25.6010706@greatiam.com> <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> Message-ID: <4B1D4E92.1040204@greatiam.com> Michael Collins wrote: > > > On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah > > wrote: > > Pardon me if this has been addressed already. > How does one go about having in the simplest instance 2 servers > registering with each other on startup whereby the users registering > would be able to call each other. > The 2 servers are in different domains. > > Thanks. > > > Are the two servers in different locations? Different LANs? Is NAT > involved? Just checking. Really this is just a matter of loading the > default config on each machine and then making some decisions about > the dialplan: do you want prefix dialing so that you can have ext 1000 > at both locations or do you want to have something like 1000~1099 at > location A and 1100~1199 at location B? From there it's just a matter > of creating the gateways on each machine and adding a dialplan entry > to handle the routing. > -MC > Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC Thanks . From msc at freeswitch.org Mon Dec 7 10:56:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:56:21 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> Message-ID: <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user you might just need to set the user so that the vars become available on the leg you're processing. -MC On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > That's exactly what I want to do and that was the first thing I tried, > but nothing is passed to the script. > > In a case like this, what defines if variable smsnumber is taken from > the A path or B path? (The A path does not have smsnumber defined) > > On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins > wrote: > > > > > > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith > > wrote: > >> > >> Hi! > >> > >> How can I access the variables that are defined in a users xml file? > >> > >> For example, say user 1000 has a variable called smsnumber, as defined > >> below: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> How can I use variable ${smsnumber} in a dialplan to run a perl script > >> using ? > >> > > > > Do you just want to pass the value in smsnumber to the sms.pl script? > Have > > you tried this? > > > > > > > > -MC > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/53615f4b/attachment.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 11:01:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 19:01:59 -0000 Subject: [Freeswitch-users] no hangup on B leg Message-ID: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/9c947a67/attachment-0001.html From mcampbellsmith at gmail.com Mon Dec 7 11:09:55 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 06:09:55 +1100 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> Message-ID: <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> Perfect... works like a charm. Thanks Mike. On Tue, Dec 8, 2009 at 5:56 AM, Michael Collins wrote: > Check out > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user > > you might just need to set the user so that the vars become available on the > leg you're processing. > -MC > > On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> That's exactly what I want to do and that was the first thing I tried, >> but nothing is passed to the script. >> >> In a case like this, what defines if variable smsnumber is taken from >> the A path or B path? (The A path does not have smsnumber defined) >> >> On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins >> wrote: >> > >> > >> > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith >> > wrote: >> >> >> >> Hi! >> >> >> >> How can I access the variables that are defined in a users xml file? >> >> >> >> For example, say user 1000 has a variable called smsnumber, as defined >> >> below: >> >> >> >> >> >> ? >> >> ? ? >> >> ? ? ? >> >> ? ? >> >> ? ? >> >> ? ? ? >> >> ? ? >> >> ? >> >> >> >> >> >> How can I use variable ${smsnumber} in a dialplan to run a perl script >> >> using ? >> >> >> > >> > Do you just want to pass the value in smsnumber to the sms.pl script? >> > Have >> > you tried this? >> > >> > >> > >> > -MC >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Dec 7 11:11:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 11:11:36 -0800 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: References: Message-ID: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi all, > > > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging > up both legs on a bridge, now today, with nothing changed, I?m not seeing a > hangup of the b leg at all. > > > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls > hangup just fine. Before when I had an issue with the B leg not closing the > bridge, I was at least getting a hangup event, now it?s not being fired. > Does anyone have an idea what might be causing this? > > > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/6ab3e27e/attachment.html From Prometheus001 at gmx.net Mon Dec 7 11:18:27 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 07 Dec 2009 20:18:27 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B1AC410.9050201@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> <4B1AC410.9050201@gmx.net> Message-ID: <4B1D5503.8010308@gmx.net> Hello, i now changed the $${domain} name of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Best regards Peter Peter P GMX schrieb: > Hello Anthony, > > I did some checks today > Here is how the phones are registered: > > mysql> select sip_host, presence_hosts, server_user,server_host, > hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip_host | presence_hosts | server_user | server_host | > hostname | sip_realm | mwi_user | mwi_host | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | > sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > IPs are: > 10.11.12.1 sip1.mydomain.com (common cluster IP) > 10.11.12.2 sip11.mydomain.com > 10.11.12.3 sip12.mydomain.com (not used at this point) > > XML-Curl for the directory is: > >
> > > > > > > > > value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > >
> > > > The internal profile has the following alias: > > > > > > > With $${domain} being sip11.mydomain.com > > Phones are registering to sip1.mydomain.com, Voicemail works, but MWI > does not. Any hint what I should change to make this work? > > Best regards > Peter > > Anthony Minessale schrieb: > >> based on your example past >> >> sip1.mydomain.com is the domain in the >> packet and thus the profile should have an alias for this. >> Then the user must reside in your sip db with the user 200 and domain >> sip1.mydomain.com >> >> if you dont have this consider the force-register-domain and >> force-register-db-domain to normalize the host names. >> >> >> On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale >> > wrote: >> >> Did you check the 2 replies that told you you need aliases in your >> sofia profile to translate the domain found in your >> message_waiting to the right profile? Both Brian and Mike >> answered you. >> >> >> >> >> >> On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX >> > wrote: >> >> I tried now with phones directly attached to the freeswitch >> (without an >> OpenSIPS in between). I also added the alias. But the >> behaviour is as >> before: >> No notify message from freeswitch, neither after register nor >> after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on >> the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > From anthony.minessale at gmail.com Mon Dec 7 11:39:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 13:39:52 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> try rerunning the ./bootstrap.sh On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards wrote: > When I got the latest trunk the make gets an error. Should I perhaps > disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. > Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 7:44 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs > without SDP. I will try to update to the latest trunk today and capture > traces as Anthony described. If I can't do it today, it might be at the end > of the week. > > Best Regards, > Jerry > > > ------------------------------ > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Saturday, December 05, 2009 7:30 PM > *To:* Jerry Richards > *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > > Why are you changing the 3pcc setting, is this an invite with no sdp? > you need to take a trace from FS. > > 1) update to latest trunk first so line number match up. > 2) issue these commands > > sofia profile internal siptrace on > console loglevel debug > > save the output and put it on pastebin http://pastebin.freeswitch.org > > > > > On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards > wrote: > >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/2e81a903/attachment-0001.html From msc at freeswitch.org Mon Dec 7 11:45:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 11:45:05 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> Message-ID: <87f2f3b90912071145o2bb416fasfcd278c766347f47@mail.gmail.com> On Mon, Dec 7, 2009 at 11:09 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Perfect... > > > > works like a charm. > "Another satisfied customer!" :P -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ac321d49/attachment.html From anthony.minessale at gmail.com Mon Dec 7 11:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 13:59:13 -0600 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <191c3a030912071159o140740b7y7214ac022cfdffc2@mail.gmail.com> also, don't use 1.0.4, please us the latest SVN or last svn snapshot at the very least. On Mon, Dec 7, 2009 at 12:34 PM, Kendall Stauffer wrote: > Any direction on where to start would be appreciated. I am trying to get > freepbx working with this, and everything works (I think) except esl > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Monday, December 07, 2009 1:10 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] esl for Mac OS X 10.4 > > > > The build system for libesl and everything below that won't work 100% on > the mac just yet. You have to make some changes to how its linked and > you'll have to compile php yourself to get everything in there properly. > The perl one however is much easier to fix. > > > > -SOLINK=-shared -Xlinker -x > > +SOLINK=-dynamiclib -Xlinker -x > > > > > > Thats all you usually fix for the mac. > > > > > > /b > > > > > > > > On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > > > > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get esl to > compile. My version is earlier than the snow leopard that is mentioned in > the general install docs, and I have tried it with and without the compiler > flags in the freewswtch installation -> MAC os X. > > I have also googled this, and don?t see what I am doing wrong. Anybody > there that can help? > > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" > CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE > -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc > -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > > _main > > __convert_to_string > > __efree > > __emalloc > > __estrndup > > __zend_get_parameters_array_ex > > __zend_list_find > > __zval_copy_ctor > > _compiler_globals > > _convert_to_long > > _zend_error > > _zend_get_constant > > _zend_hash_find > > _zend_register_list_destructors_ex > > _zend_register_long_constant > > _zend_register_resource > > _zend_rsrc_list_get_rsrc_type > > _zend_wrong_param_count > > collect2: ld returned 1 exit status > > make[1]: *** [ESL.so] Error 1 > > make: *** [phpmod] Error 2 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/dd9a5c4b/attachment.html From sklein at singular.com Mon Dec 7 12:19:01 2009 From: sklein at singular.com (Steve Klein) Date: Mon, 7 Dec 2009 12:19:01 -0800 Subject: [Freeswitch-users] Database suggestions/pointers/? In-Reply-To: <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> References: <053d01ca76af$c311d2c0$49357840$@com> <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> Message-ID: <06c901ca777a$845ec800$8d1c5800$@com> Thanks for the suggestions. We'll explore. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Uckun Sent: Sunday, December 06, 2009 5:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Database suggestions/pointers/? On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein wrote: > Greetings. We need to add database access to an IVR application we are > prototyping. Based on FS best practice suggestions, we are using Lua for > the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite > would be a recommended approach. However, we cant find any examples of this > combo anywhere. So, what is the best practice scripting + database > recommendation for a high-volume database-driven FS app? > I would suggest you take a look at freeswitcher (http://github.com/bougyman/freeswitcher). The good thing is that it's ruby and therefore you can use any database compatible with ruby (that's all of them pretty much). You can also use an ORM of your choice or if you don't want to use an ORM you can use the amazingly fantastic sequel library. Being ruby it will run outside of the freeswitch memory space and you will have to use the inbound/outbound socket API. That may be a good thing if you want to separate your database and IVR logic from the machine running your freeswitch. Ruby is pretty easy to pick up if you don't know it and there are a wealth of libraries if you want to do other things like connect to web sites, manipulate XML, etc. There is also a liverpie http://github.com/jsgoecke/liverpie which is more of a proxy thing you can interface with any language. I am sure lua is nice but it seems like people are having some problems with ODBC, memory leaks etc when it comes to databases. If you go a ruby library that all goes away. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sklein at singular.com Mon Dec 7 12:19:48 2009 From: sklein at singular.com (Steve Klein) Date: Mon, 7 Dec 2009 12:19:48 -0800 Subject: [Freeswitch-users] lua+sqlite example? In-Reply-To: <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> References: <04a201ca7623$2c0b2020$84216060$@com> <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> Message-ID: <06ca01ca777a$a04125e0$e0c371a0$@com> Thanks. We'll look at that. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, December 07, 2009 9:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] lua+sqlite example? yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein wrote: Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.426 / Virus Database: 270.14.83/2529 - Release Date: 12/07/09 07:33:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ec119444/attachment-0001.html From mailinglist at fribert.dk Mon Dec 7 12:50:39 2009 From: mailinglist at fribert.dk (mailinglist) Date: Mon, 07 Dec 2009 21:50:39 +0100 Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through Message-ID: <4B1D78AF020000E1000002BC@mail.fribert.dk> Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia at 10.11.12.25:5060 BIND-URL sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1001 at 10.11.12.25 Contact: "1001" Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID: OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1002 at 10.11.12.25 Contact: "1002" Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 ================================================================================================= And... sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia at 87.61.18.196:5080 BIND-URL sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 Registrations: ================================================================================================= ================================================================================================= In my Dialplan I've created these two entries: ----- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- and ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$1 I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. If I try and make a local call from 1001 to 1002 it says 2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001 at 1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0d73912f/attachment.html From erandr-junk at usa.net Mon Dec 7 12:58:55 2009 From: erandr-junk at usa.net (eaf) Date: Mon, 7 Dec 2009 12:58:55 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <26684048.post@talk.nabble.com> What do you want me to check while running these tests? Sound quality (it's good now even with original 1.0.4). Or CPU utilization? It's Debian 4. Anthony Minessale-2 wrote: > > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least > as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly > can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm. We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using. I am > curious > what would happen if you install Kristian's astlinux on one of your > devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do > these > systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer > thread > which will make all the switch_cond_next share a 1ms conditional instead > of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others > work > better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > >> >> Here is what I found... >> >> I tried high-priority scheduling as per your suggestion, reniced the >> program >> explicitly, rewrote timer thread to sleep on cond. variable and activate >> only when there are timers and only when the timer actually had to be >> clicked, turned off SQL thread and removed polling from sofia profile >> thread. >> >> That pretty much eliminated all idle 1ms sleepers that were there except >> for >> three in sofia itself (su_epoll_port). And when I was about to be happy, >> I >> found that two outgoing calls through my VOIP providers when bridged >> together showed terrible distortions. I undid all my changes, tried >> 1.0.4, >> trunk (noticed btw that when I bridge two calls via loopback in JS in the >> trunk I must keep JS running, or the calls get terminated - NOT the same >> as >> in 1.0.4 where exitting JS left calls running), got pretty much the same >> sad >> results. At the same time calls bridged by freeswitch between LAN and any >> of >> the VOIP providers behaved just fine. And calls bridged by Asterisk any >> way >> were fine too. So that pretty much looked like the end of the freeswitch >> trials for me. >> >> But then I timed your code, mine and found that all those 1ms sleeps that >> your timer thread was doing (and all those pollers were doing as well) >> were >> actually 4ms sleeps because you know what unless kernel is configured >> with >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms >> (HZ=100). Mine was 250. >> >> This actually meant that the original timer thread was firing once, >> sleeping >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing >> 4 >> times back-to-back, etc. It was still firing 20ms timers on time, but >> 30ms >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus >> whoever >> relied on runtime.reference or switch_micro_time_now() were kind of >> screwed >> because both were running jumpy. Plus whoever assumed that >> apr_sleep(1000) >> or cond_yield() was sleeping for 1ms were also in for a surprise. It felt >> satisfying to find that, however it didn't explain why the same >> distortions >> were observed with rewritten timer thread and disabled RTP timers. >> >> Anyway, I sighed (pretty much like you) and recompiled the kernel with >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes >> south, >> you need to hook up serial console and see what the heck went wrong. >> >> That eliminated distortions, ha! But made freeswitch more CPU hungry. Now >> the remaining 1ms threads sitting in sofia epoll were really polling for >> 1ms, not 4, and freeswitch was consistently sitting in the first line of >> the >> top chart showing 3% CPU utilization when idle. >> >> Don't know whether it's because of the remaining epolls in sofia or >> whether >> it's because there are still some threads left in freeswitch that I >> neglected to change because they were sleeping with 100ms interval, so I >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. >> >> Btw, compared with Asterisk, the latter is not even visible on the first >> top's screen and spends 1% CPU when bridging two G711 calls and recording >> them to disk. >> >> So, at this time I have both original Asterisk and FS setups running. One >> is >> seemless but clumsy in configuration, the other one is neat and stylish >> but >> too preoccupied with smth... Should I look into sofia epollers? That's >> kind >> of deep in the code. Or should I just stick with Asterisk? >> >> >> >> >> >> Anthony Minessale-2 wrote: >> > >> > There is another user here with a 300mhz box. I am willing to >> investigate >> > this improved performance for weak devices but I need to do it in a >> sane >> > cross-platform way. >> > >> > >> > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman >> > > >> wrote: >> > >> >> A word to the wise to the general FreeSWITCH community: If Anthony >> >> Minessale suggests that you try to do any number of things, it's a >> very >> >> good idea to try all those ideas before continuing on. I've known >> him, >> >> MikeJ, and bkw for several years, and they almost always have very >> good >> >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely >> >> frustrating to try to help people out who won't try the provided >> >> suggestions first. >> >> >> >> And note directly to "eaf" - bogomips is quite possibly the least >> >> significant bit of data about a cpu that you will get out of >> >> /proc/cpuinfo... The name itself - bogo, means bogus. >> >> http://en.wikipedia.org/wiki/Bogomips >> >> >> >> -Yossi >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26684048.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Dec 7 13:06:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 21:06:29 -0000 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/8df13f49/attachment.html From anthony.minessale at gmail.com Mon Dec 7 13:29:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 15:29:15 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26684048.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> <26684048.post@talk.nabble.com> Message-ID: <191c3a030912071329h374a1efcl240ae6e3e45e7810@mail.gmail.com> Both, if it always sounds ok then I guess CPU usage. On Mon, Dec 7, 2009 at 2:58 PM, eaf wrote: > > What do you want me to check while running these tests? Sound quality (it's > good now even with original 1.0.4). Or CPU utilization? > > It's Debian 4. > > > Anthony Minessale-2 wrote: > > > > Did you do each thing alone too to tell the difference? > > -hp alone, disable monotonic alone (i did not see you mention the disable > > monotonic) > > > > as for your 4ms thing, yes we require high resolution timing, if we ask > to > > sleep 1000 microseconds that is what we need it to sleep for or at least > > as > > close as possible, and the main reason that thread is never sleeping is > > because you can't actually count on it to run every 1ms but you mostly > > can. > > Hence the whole philosophy on only making 1 thread run hot all the time > to > > ensure that the rest don't have to repeat the same algorithm. We focus > on > > high end performance this was the point of your experimentation because > we > > will need to use a compile time defines and other logic to make it more > > efficient on your platform, a platform which we are not using. I am > > curious > > what would happen if you install Kristian's astlinux on one of your > > devices, > > i think you should also compare the kernel versions. > > > > > > What OS are you running anyway? > > > > Here are some more things to try (running plain trunk with no mods) do > > these > > systematically each alone and all together with/without -hp or disable > > monotonic etc to see what different combos create > > > > comment out this line (line 10) > > #define DISABLE_1MS_COND > > > > rebuild, this tells it to run a conditional at 1ms in the same timer > > thread > > which will make all the switch_cond_next share a 1ms conditional instead > > of > > doing microsleeps > > > > next > > > > some kernels/devices work better using select(0) for sleep where others > > work > > better using usleep. > > comment out line 109 > > apr_sleep(t); > > > > and try > > usleep(t) > > > > also mac works better using nanosleep so you could try changing it so it > > uses the code starting at 101 instead. > > > > > > also your claim about JS should be investigated because I do not think it > > should be the case. > > but you may want to move this to a jira http://jira.freeswitch.org > > > > As for the asterisk comparison, > > not sure how to answer you, that's your decision. > > > > > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > > >> > >> Here is what I found... > >> > >> I tried high-priority scheduling as per your suggestion, reniced the > >> program > >> explicitly, rewrote timer thread to sleep on cond. variable and activate > >> only when there are timers and only when the timer actually had to be > >> clicked, turned off SQL thread and removed polling from sofia profile > >> thread. > >> > >> That pretty much eliminated all idle 1ms sleepers that were there except > >> for > >> three in sofia itself (su_epoll_port). And when I was about to be happy, > >> I > >> found that two outgoing calls through my VOIP providers when bridged > >> together showed terrible distortions. I undid all my changes, tried > >> 1.0.4, > >> trunk (noticed btw that when I bridge two calls via loopback in JS in > the > >> trunk I must keep JS running, or the calls get terminated - NOT the same > >> as > >> in 1.0.4 where exitting JS left calls running), got pretty much the same > >> sad > >> results. At the same time calls bridged by freeswitch between LAN and > any > >> of > >> the VOIP providers behaved just fine. And calls bridged by Asterisk any > >> way > >> were fine too. So that pretty much looked like the end of the freeswitch > >> trials for me. > >> > >> But then I timed your code, mine and found that all those 1ms sleeps > that > >> your timer thread was doing (and all those pollers were doing as well) > >> were > >> actually 4ms sleeps because you know what unless kernel is configured > >> with > >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > >> (HZ=100). Mine was 250. > >> > >> This actually meant that the original timer thread was firing once, > >> sleeping > >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing > >> 4 > >> times back-to-back, etc. It was still firing 20ms timers on time, but > >> 30ms > >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus > >> whoever > >> relied on runtime.reference or switch_micro_time_now() were kind of > >> screwed > >> because both were running jumpy. Plus whoever assumed that > >> apr_sleep(1000) > >> or cond_yield() was sleeping for 1ms were also in for a surprise. It > felt > >> satisfying to find that, however it didn't explain why the same > >> distortions > >> were observed with rewritten timer thread and disabled RTP timers. > >> > >> Anyway, I sighed (pretty much like you) and recompiled the kernel with > >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes > >> south, > >> you need to hook up serial console and see what the heck went wrong. > >> > >> That eliminated distortions, ha! But made freeswitch more CPU hungry. > Now > >> the remaining 1ms threads sitting in sofia epoll were really polling for > >> 1ms, not 4, and freeswitch was consistently sitting in the first line of > >> the > >> top chart showing 3% CPU utilization when idle. > >> > >> Don't know whether it's because of the remaining epolls in sofia or > >> whether > >> it's because there are still some threads left in freeswitch that I > >> neglected to change because they were sleeping with 100ms interval, so I > >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) > >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. > >> > >> Btw, compared with Asterisk, the latter is not even visible on the first > >> top's screen and spends 1% CPU when bridging two G711 calls and > recording > >> them to disk. > >> > >> So, at this time I have both original Asterisk and FS setups running. > One > >> is > >> seemless but clumsy in configuration, the other one is neat and stylish > >> but > >> too preoccupied with smth... Should I look into sofia epollers? That's > >> kind > >> of deep in the code. Or should I just stick with Asterisk? > >> > >> > >> > >> > >> > >> Anthony Minessale-2 wrote: > >> > > >> > There is another user here with a 300mhz box. I am willing to > >> investigate > >> > this improved performance for weak devices but I need to do it in a > >> sane > >> > cross-platform way. > >> > > >> > > >> > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > >> > >> >> wrote: > >> > > >> >> A word to the wise to the general FreeSWITCH community: If Anthony > >> >> Minessale suggests that you try to do any number of things, it's a > >> very > >> >> good idea to try all those ideas before continuing on. I've known > >> him, > >> >> MikeJ, and bkw for several years, and they almost always have very > >> good > >> >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > >> >> frustrating to try to help people out who won't try the provided > >> >> suggestions first. > >> >> > >> >> And note directly to "eaf" - bogomips is quite possibly the least > >> >> significant bit of data about a cpu that you will get out of > >> >> /proc/cpuinfo... The name itself - bogo, means bogus. > >> >> http://en.wikipedia.org/wiki/Bogomips > >> >> > >> >> -Yossi > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > -- > >> > Anthony Minessale II > >> > > >> > FreeSWITCH http://www.freeswitch.org/ > >> > ClueCon http://www.cluecon.com/ > >> > Twitter: http://twitter.com/FreeSWITCH_wire > >> > > >> > AIM: anthm > >> > MSN:anthony_minessale at hotmail.com > >> > >< > >> MSN%3Aanthony_minessale at hotmail.com > > > > >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > > > > >> > > >> > IRC: irc.freenode.net #freeswitch > >> > > >> > FreeSWITCH Developer Conference > >> > sip:888 at conference.freeswitch.org > >> > >< > >> sip%3A888 at conference.freeswitch.org > > > > >> > > >> > iax:guest at conference.freeswitch.org/888 > >> > > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> > > > > >> > > >> > pstn:213-799-1400 > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26684048.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/625b8007/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:02:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:02:42 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call Message-ID: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/33d48ca5/attachment.html From brian at freeswitch.org Mon Dec 7 14:12:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 16:12:59 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <6350BFD5-D6E4-42CB-AB46-7677F1B4D537@freeswitch.org> session:execute("start_dtmf"); /b On Dec 7, 2009, at 4:02 PM, Nik Middleton wrote: > Hi > > Is it possible to trap on DTMF on a bridged call within an LUA > script? I?ve tried setting the gateway to use inband, but no joy. > It looks like I could use start_dtmf, but I can?t see how to launch > this within LUA > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/af59690d/attachment.html From anthony.minessale at gmail.com Mon Dec 7 14:15:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 16:15:16 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> session:execute("start_dtmf"); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > > > Regards, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/adf1a629/attachment.html From msc at freeswitch.org Mon Dec 7 14:18:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 14:18:20 -0800 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ad082b09/attachment.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:56:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:56:16 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> References: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> Message-ID: Once the call is bridged, while I can see an inband DTMF event being generated, it doesn't call my hook unfortuneately function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '*' then session:hangup(); return true; end session:execute("start_dtmf"); session:execute("bridge",bridgestring ); Am I missing something? Before the bridge, the oninput function works fine Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 22:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call session:execute("start_dtmf"); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0ce7a20d/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:59:03 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:59:03 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: Can this be done in an lua script? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/23b7bc42/attachment.html From Prometheus001 at gmx.net Mon Dec 7 15:12:29 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 08 Dec 2009 00:12:29 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B1AC410.9050201@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> <4B1AC410.9050201@gmx.net> Message-ID: <4B1D8BDD.7040505@gmx.net> Hello, i now changed the $${domain} of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Thanks to all for your support. Best regards Peter Peter P GMX schrieb: > Hello Anthony, > > I did some checks today > Here is how the phones are registered: > > mysql> select sip_host, presence_hosts, server_user,server_host, > hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip_host | presence_hosts | server_user | server_host | > hostname | sip_realm | mwi_user | mwi_host | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | > sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > IPs are: > 10.11.12.1 sip1.mydomain.com (common cluster IP) > 10.11.12.2 sip11.mydomain.com > 10.11.12.3 sip12.mydomain.com (not used at this point) > > XML-Curl for the directory is: > >
> > > > > > > > > value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > >
> > > > The internal profile has the following alias: > > > > > > > With $${domain} being sip11.mydomain.com > > Phones are registering to sip1.mydomain.com, Voicemail works, but MWI > does not. Any hint what I should change to make this work? > > Best regards > Peter > > Anthony Minessale schrieb: > >> based on your example past >> >> sip1.mydomain.com is the domain in the >> packet and thus the profile should have an alias for this. >> Then the user must reside in your sip db with the user 200 and domain >> sip1.mydomain.com >> >> if you dont have this consider the force-register-domain and >> force-register-db-domain to normalize the host names. >> >> >> On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale >> > wrote: >> >> Did you check the 2 replies that told you you need aliases in your >> sofia profile to translate the domain found in your >> message_waiting to the right profile? Both Brian and Mike >> answered you. >> >> >> >> >> >> On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX >> > wrote: >> >> I tried now with phones directly attached to the freeswitch >> (without an >> OpenSIPS in between). I also added the alias. But the >> behaviour is as >> before: >> No notify message from freeswitch, neither after register nor >> after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on >> the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > From larclap at yahoo.com Mon Dec 7 15:19:42 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 7 Dec 2009 15:19:42 -0800 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: <010001ca7793$c1e7c410$45b74c30$@com> It can. I use it like: session:execute("bind_meta_app", "1 b s execute_extension::dx XML features"); session:execute("bind_meta_app", "2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav"); session:execute("bind_meta_app", "3 b s execute_extension::cf XML features"); From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Monday, December 07, 2009 2:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call Can this be done in an lua script? Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0dd25521/attachment.html From anthony.minessale at gmail.com Mon Dec 7 15:21:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 17:21:22 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Can this be done in an lua script? > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 07 December 2009 22:18 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Trapping dtmf on bridged call > > > > > > On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever > you want to have happen. Check it out: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app > > The Local_Extension in the default.xml dialplan file has a few examples of > using this tool. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/48514461/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 15:32:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 23:32:25 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> Message-ID: Yes I did, is it possible mod_vmd is interering? It's stopped before I call the start_dtmf function session:setHangupHook("myHangupHook", "blah") session:setInputCallback("onInput"); session:execute("vmd","start"); if (session:ready() == false) then freeswitch.consoleLog("info", " : Call Failed!!!\n"); end session:answer(); ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton wrote: Can this be done in an lua script? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/844a4375/attachment.html From chris at fowler.cc Mon Dec 7 16:46:11 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 7 Dec 2009 19:46:11 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <1260233114.13588.1348952845@webmail.messagingengine.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> Message-ID: <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> Skype have opened their beta program up to all comers. http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent Three lines in a sip_profile make FreeSWITCH talk nicely; but using the PCMU codec. Any progress on SILK native support? Last I saw was discussion back in September with Brian lamenting that Skype was hard to work with on this. I know I could use mod_skypiax; but having a native solution would be one less IT headache. Thx, Chris. From brian at freeswitch.org Mon Dec 7 17:27:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 19:27:46 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> Message-ID: <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> They have yet to type "make" on a 64bit box and build us a binary that is 64bit. Chances are they mucked it up like the BroadVoice codecs were and it just won't work on 64bit just yet... if they would just give us the src we could be done in under two days with it I suspect. /b On Dec 7, 2009, at 6:46 PM, Chris Fowler wrote: > Skype have opened their beta program up to all comers. > http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent > > Three lines in a sip_profile make FreeSWITCH talk nicely; but using > the > PCMU codec. > > Any progress on SILK native support? Last I saw was discussion back in > September with Brian lamenting that Skype was hard to work with on > this. > > I know I could use mod_skypiax; but having a native solution would be > one less IT headache. > > Thx, Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4a531d2c/attachment.html From jason at jasonjgw.net Mon Dec 7 17:39:38 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 8 Dec 2009 12:39:38 +1100 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> Message-ID: <20091208013938.GA4976@jdc.jasonjgw.net> Brian West wrote: > They have yet to type "make" on a 64bit box and build us a binary > that is 64bit. Chances are they mucked it up like the BroadVoice > codecs were and it just won't work on 64bit just yet... if they > would just give us the src we could be done in under two days with > it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) From brian at freeswitch.org Mon Dec 7 17:50:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 19:50:08 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <20091208013938.GA4976@jdc.jasonjgw.net> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> Message-ID: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> it I suspect. > > Given that they released the codec specification, perhaps someone is > writing > an independent C implementation? (Not that I'm much interested, > but...) From mctch at yahoo.com Mon Dec 7 18:05:52 2009 From: mctch at yahoo.com (Mark Crane) Date: Mon, 7 Dec 2009 18:05:52 -0800 (PST) Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1D78AF020000E1000002BC@mail.fribert.dk> Message-ID: <659603.29094.qm@web56408.mail.re3.yahoo.com> Question ---------------------------------------------- I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- Since you moved the internal profile to the lan ip address you can go ahead and dump the lan profile. Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer --- On Mon, 12/7/09, mailinglist wrote: From: mailinglist Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Monday, December 7, 2009, 1:50 PM Hi All ? Ok, so next episode in the saga of getting this monster of the ground :-) ? I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. ? If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] ? I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? ? ? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name??????????????????? internal Domain Name???????????? N/A DBName????????????????? sofia_reg_internal Pres Hosts Dialplan??????????????? XML Context???????????????? public Challenge Realm???????? auto_from RTP-IP????????????????? 10.11.12.25 Ext-RTP-IP????????????? 10.11.12.25 SIP-IP????????????????? 10.11.12.25 Ext-SIP-IP????????????? 10.11.12.25 URL???????????????????? sip:mod_sofia at 10.11.12.25:5060 BIND-URL??????????????? sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC????????????? local_stream://moh OUTBOUND-PROXY????????? N/A CODECS????????????????? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT?????????????? 101 DTMF-MODE?????????????? rfc2833 CNG???????????????????? 13 SESSION-TO????????????? 0 MAX-DIALOG????????????? 0 NOMEDIA???????????????? false LATE-NEG??????????????? false PROXY-MEDIA???????????? false AGGRESSIVENAT?????????? false STUN-ENABLED??????????? true STUN-AUTO-DISABLE?????? false CALLS-IN??????????????? 10 FAILED-CALLS-IN???????? 5 CALLS-OUT?????????????? 0 FAILED-CALLS-OUT??????? 0 ? Registrations: ================================================================================================= Call-ID:??????? MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User:?????????? 1001 at 10.11.12.25 Contact:??????? "1001" Agent:????????? X-Lite release 1103k stamp 53621 Status:???????? Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host:?????????? firewall.fribert.dk IP:???????????? 10.11.12.145 Port:?????????? 59650 Auth-User:????? 1001 Auth-Realm:???? 10.11.12.25 ? Call-ID:??????? OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User:?????????? 1002 at 10.11.12.25 Contact:??????? "1002" Agent:????????? 3CXVoipPhone 3.1.6288.0 Status:???????? Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host:?????????? firewall.fribert.dk IP:???????????? 10.11.12.195 Port:?????????? 4117 Auth-User:????? 1002 Auth-Realm:???? 10.11.12.25 ? ================================================================================================= ? And... ? sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name??????????????????? external Domain Name???????????? N/A DBName????????????????? sofia_reg_external Pres Hosts Dialplan??????????????? XML Context???????????????? public Challenge Realm???????? auto_to RTP-IP????????????????? 87.61.18.196 Ext-RTP-IP????????????? 87.61.18.196 SIP-IP????????????????? 87.61.18.196 Ext-SIP-IP????????????? 87.61.18.196 URL???????????????????? sip:mod_sofia at 87.61.18.196:5080 BIND-URL??????????????? sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC????????????? local_stream://moh OUTBOUND-PROXY????????? N/A CODECS????????????????? PCMU,PCMA,GSM TEL-EVENT?????????????? 101 DTMF-MODE?????????????? rfc2833 CNG???????????????????? 13 SESSION-TO????????????? 0 MAX-DIALOG????????????? 0 NOMEDIA???????????????? false LATE-NEG??????????????? false PROXY-MEDIA???????????? false AGGRESSIVENAT?????????? false STUN-ENABLED??????????? true STUN-AUTO-DISABLE?????? false CALLS-IN??????????????? 0 FAILED-CALLS-IN???????? 0 CALLS-OUT?????????????? 2 FAILED-CALLS-OUT??????? 2 ? Registrations: ================================================================================================= ================================================================================================= ? ? In my Dialplan I've created these two entries: ? ----- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- ? and ? ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$1 ? I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. ? If I try and make a local call from 1001 to 1002 it says ?2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] ? As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. ? Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001 at 1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here??? ? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0411e0d7/attachment-0001.html From djbinter at yahoo.com Mon Dec 7 18:29:12 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 18:29:12 -0800 (PST) Subject: [Freeswitch-users] Zombie Records in core db Message-ID: <151010.8763.qm@web37501.mail.mud.yahoo.com> We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from "show calls" that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those "zombied" records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/306f7071/attachment.html From pete at privateconnect.com Mon Dec 7 19:16:47 2009 From: pete at privateconnect.com (Pete Mueller) Date: Mon, 7 Dec 2009 20:16:47 -0700 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> Message-ID: <002401ca77b4$e0d64d80$a282e880$@com> I had this featured requested of me a few months ago. Bind_meta_app does work, but requires two tones, the "*" plus an additional digit. I needed to perform a task on the "*". I re-wrote the bind_meta_app handler so that if you attached a instruction to what would be "**" hitting a single "*" would active it. Kind of a hack, but if no one comes up with a more elegant way, I could provide a .patch file that did the necessary changes. I'd love an all-LUA method, or something that could use an existing InputCallback routine, but this worked for my immediate need. -pete From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Monday, December 07, 2009 4:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call Yes I did, is it possible mod_vmd is interering? It's stopped before I call the start_dtmf function session:setHangupHook("myHangupHook", "blah") session:setInputCallback("onInput"); session:execute("vmd","start"); if (session:ready() == false) then freeswitch.consoleLog("info", " : Call Failed!!!\n"); end session:answer(); _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton wrote: Can this be done in an lua script? Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/5a732875/attachment.html From dujinfang at gmail.com Mon Dec 7 19:33:27 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 8 Dec 2009 11:33:27 +0800 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <23f91030912071933w231c61e6gcd4e2aa555f2794d@mail.gmail.com> I also have this problem on a trunk version more than 1000 revisions behind, so I think the best way is to upgrade to trunk and report this again if still have problem. 2009/12/8 DJB : > We have FreeSWITCH Version 1.0.4 (exported) running at a high volume > traffic. ?I normally check the concurrent calls by looking at the number of > sessions from status command. ?However, the number of concurrent calls in FS > is normally higher than it's supposed to be after we ran traffic for about a > week. ?Thus, I routed the traffic away from the FS and found out from "show > calls" that there were so many old calls from previous days. ?We are running > a pass-thru traffic in signaling only. ?I wonder whether there is a way to > have those "zombied" records clean up automatically. ?Also, what should I do > to prevent this problem? > Thank you, > Dorn B. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Dec 7 19:41:03 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 7 Dec 2009 19:41:03 -0800 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <580F5165-09AB-44DF-B857-E9B1A43D076D@freeswitch.org> Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk or at least pre8 when it's available. -MC Sent from my iPhone On Dec 7, 2009, at 6:29 PM, DJB wrote: > We have FreeSWITCH Version 1.0.4 (exported) running at a high volume > traffic. I normally check the concurrent calls by looking at the > number of sessions from status command. However, the number of > concurrent calls in FS is normally higher than it's supposed to be > after we ran traffic for about a week. Thus, I routed the traffic > away from the FS and found out from "show calls" that there were so > many old calls from previous days. We are running a pass-thru > traffic in signaling only. I wonder whether there is a way to have > those "zombied" records clean up automatically. Also, what should I > do to prevent this problem? > > Thank you, > Dorn B. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/dc872afe/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 7 19:47:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 21:47:01 -0600 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <191c3a030912071947q3dd98adbn320b2a1b7f1b25bc@mail.gmail.com> For starters, try using the latest svn snapshot. Your version is 6 months old and several thousand revs old. On Dec 7, 2009 8:34 PM, "DJB" wrote: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from "show calls" that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those "zombied" records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/91a99bbe/attachment.html From mailinglist at fribert.dk Mon Dec 7 22:05:31 2009 From: mailinglist at fribert.dk (mailinglist) Date: Tue, 08 Dec 2009 07:05:31 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <659603.29094.qm@web56408.mail.re3.yahoo.com> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> Message-ID: <4B1DFABC020000E1000002C2@mail.fribert.dk> Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/20d25408/attachment.html From yehavi.bourvine at gmail.com Mon Dec 7 22:50:11 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Dec 2009 08:50:11 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> <4B1C031B.8060906@metik.com> <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> Message-ID: Hello all, *debug voip rtp session named-event*s shows that it receives and understands the DTMFs, but it does not send them to the PSTN (sends only those received via INFO). I haveto find some time and go to the remote site to update to the latest IOS... I will update after this has been done. Regards, __Yehavi: 2009/12/6 Anthony Minessale > Some more bad news for you, info dtmf spec has expired and has been > abandoned. Wait till you see what they did accept instead...... > > On Dec 6, 2009 1:22 PM, "Metik" wrote: > > Unless the IOS you are running is extremely buggy, "debug voip ccapi" > commands should not provide you with that detail, what you really want > to use is "debug voip rtp session named-event". > > Normal SIP-to-PSTN calls should use both a pots and voip dial peer but > DTMF relay type is determined by the voip dial peer. > > I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) > previously in the wild. Unlike some other SIP feature servers, I have > not had issues (with RFC 2833) between FS and Cisco IOS gateways. > > Although unrelated to FS or any other SIP feature server, I have seen > some issues when multple dtmf relay types are left enabled on a voip > dial peer. Also, there are some (older) IOS versions that have issues > with DTMF duration which cause digits to be misinterpreted by the > far-end (PSTN/POTS) but not ignored altogether. > > -metik > > Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik < > freeswitch-users-list at metik.com > > > > > > > You previously stated that your Cisco gateway has some "bug" that > > prevents you from us... > > > ------------------------------------------------------------------------ > > > _____________________... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e07a31a9/attachment.html From jingwei.yang at gmail.com Tue Dec 8 01:09:29 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 8 Dec 2009 17:09:29 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> Message-ID: <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> Hi Jo?o, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. Here's how I normally update FreeSwitch: *make clean && svn up && ./bootstrap.sh && ./configure && make install * If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. Regards, -Jingwei 2009/12/8 Jo?o Mesquita > Maybe, just maybe isse that make target to reconf libtiff? > > Regards, > > JM > > > On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > >> I installed libjpeg-7 following this website: >> http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And >> the previous error is replaced by a new one: >> >> gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. >> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF >> .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o >> at_interpreter.o >> at_interpreter.c: In function ???command_search???: >> at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use >> in this function) >> at_interpreter.c:5299: error: (Each undeclared identifier is reported only >> once >> at_interpreter.c:5299: error: for each function it appears in.) >> at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in >> this function) >> at_interpreter.c: In function ???at_interpreter???: >> at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in >> this function) >> make[8]: *** [at_interpreter.lo] Error 1 >> >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> However, I'm still able to start freeswitch and mod_skypiax and make skype >> calls with no problem. >> >> Regards, >> -Jingwei >> >> >> >> On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: >> >>> No, I didn't change or update the system libs. I just wanted to double >>> check whether my system has this libjpeg library. ./configure was definitely >>> executed before the source codes were rebuilt. >>> >>> Regards, >>> -Jingwei >>> >>> >>> On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: >>> >>>> Hi, >>>> >>>> That one is on your side. If you changed/updated system libs it might be >>>> worth doing another ./configure >>>> >>>> Cheers, >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >>>> >>>> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >>>> However, I encounter another one. >>>> >>>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >>>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >>>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>>> -lc >>>> ./make_at_dictionary: error while loading shared libraries: >>>> libjpeg.so.7: cannot open shared object file: No such file or directory >>>> make[8]: *** [at_interpreter_dictionary.h] Error 127 >>>> make[7]: *** [all] Error 2 >>>> make[6]: *** [all-recursive] Error 1 >>>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >>>> make[4]: *** [install] Error 1 >>>> make[3]: *** [mod_voipcodecs-install] Error 1 >>>> make[2]: *** [install-recursive] Error 1 >>>> >>>> Do you have idea about this one? >>>> >>>> Thanks! >>>> >>>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >>>> >>>>> Consider it fixed. >>>>> Committed revision 15765. >>>>> >>>>> Mathieu Rene >>>>> Avant-Garde Solutions Inc >>>>> Office: + 1 (514) 664-1044 x100 >>>>> Cell: +1 (514) 664-1044 x200 >>>>> mrene at avgs.ca >>>>> >>>>> >>>>> >>>>> >>>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>>>> >>>>> Hi Guys, >>>>> >>>>> I got a compilation error of skypiax_protocol.c with the latest version >>>>> r15764. >>>>> >>>>> Compiling skypiax_protocol.c... >>>>> *cc1: warnings being treated as errors* >>>>> skypiax_protocol.c: In function ???X11_errors_handler???: >>>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c: In function ???skypiax_send_message???: >>>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> make[5]: *** [skypiax_protocol.o] Error 1 >>>>> make[4]: *** [install] Error 1 >>>>> make[3]: *** [mod_skypiax-install] Error 1 >>>>> make[2]: *** [install-recursive] Error 1 >>>>> >>>>> I personally checked the file and it shouldn't be a merge problem. Does >>>>> anyone encounter this as well? >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/cb574ba9/attachment-0001.html From jingwei.yang at gmail.com Tue Dec 8 02:01:57 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 8 Dec 2009 18:01:57 +0800 Subject: [Freeswitch-users] OpenZap issues with incoming and outgoing calls In-Reply-To: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> References: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> Message-ID: <13529f9d0912080201s8f6db58w7fafc3a41de3739f@mail.gmail.com> Problem solved. It's due to the lack of definition in tones.conf. In case anyone else need it, here's the tone plan for Singapore. [sg] generate-dial => v=-7;%(1000,0,425) detect-dial => 425 generate-ring => v=-7;%(2000,4000,425) detect-ring => 425 generate-busy => v=-7;%(750,750,425) detect-busy => 425 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 1776.7 On Thu, Dec 3, 2009 at 5:29 PM, Jingwei Yang wrote: > Hello All, > > I have a Digium TDM400P pci card with two FXO ports installed on my linux > box. I've connected an external telephone line to the first FXO port. But I > can't either make outgoing calls or receive incoming ones. Here are my > setups, please let me know where goes wrong. > * > /etc/zaptel.conf* > > loadzone = sg > defaultzone=sg > fxsks=1,2 > > */usr/local/freeswitch/conf/zt.conf* remains unchanged > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > rxgain => 0.0 > txgain => 0.0 > > */usr/local/freeswitch/conf/openzap.conf* > > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > [span zt] > name => OpenZAP > number => 2 > fxo-channel => 2 > > */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* > > > > > > > > > > > > > > > > > > > > > > > > > I defined an extension in dialplan/default.xml to receive bridge incoming > calls to my skype instance. Frankly speaking, I'm not sure whether this > definition is correct. How should I define the expression? When I dial the > telephone number, the FS console has no response and I hear nother but busy > tones. > > > > > > > > For outgoing calls, I tried something like this: originate > openzap/1/1/xxxxxxxx &echo, while "xxxxxxxx" is my handphone number. Again, > my handphone has no response. Hopefully I've explained my situation clearly. > Please kindly enlighten where the problem might be. > > Thanks, > -Jingwei > > p.s. here is the outgoing log trace for your reference. > > > freeswitch at localhost.localdomain> originate openzap/1/1/xxxxxxxx &echo > 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 > taps for 1:1 > 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound > channel OpenZAP/1:1/xxxxxxxx > 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel > OpenZAP/1:1/xxxxxxxx [6f843194-18ce-4525-862f-f5f4e96db5eb] > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 > (OpenZAP/1:1/xxxxxxxx) State Change CS_NEW -> CS_INIT > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 > from DOWN to DIALING > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread > starting. > 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 > taps for 1:1 > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for DIALING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_INIT > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/1:1/xxxxxxxx) State INIT > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/xxxxxxxx) > State Change CS_INIT -> CS_ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/1:1/xxxxxxxx) State INIT going to sleep > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/1:1/xxxxxxxx) State ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/xxxxxxxx > CHANNEL ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 > (OpenZAP/1:1/xxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/1:1/xxxxxxxx) State ROUTING going to sleep > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA going to sleep > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_HOME ||| > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_OFFICE ||| > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_MOBILE ||| > 2009-12-03 17:21:34.684709 [DEBUG] ozmod_analog.c:340 Changing state on 1:1 > from DIALING to BUSY > 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for BUSY > 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:579 Changing state on 1:1 > from BUSY to DOWN > 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for DOWN > 2009-12-03 17:21:34.724706 [DEBUG] mod_openzap.c:1334 got FXO sig 1:1 > [STOP] > 2009-12-03 17:21:34.724706 [NOTICE] mod_openzap.c:1352 Hangup > OpenZAP/1:1/xxxxxxxx [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] > 2009-12-03 17:21:34.724706 [DEBUG] switch_channel.c:1912 Send signal > OpenZAP/1:1/xxxxxxxx [KILL] > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_HANGUP > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > 2009-12-03 17:21:34.724706 [DEBUG] zap_io.c:1234 channel done 1:1 > 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:766 ANALOG CHANNEL 1:1 > thread ended. > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:486 > (OpenZAP/1:1/xxxxxxxx) State HANGUP > API CALL [originate(openzap/1/1/xxxxxxxx &echo)] output: > -ERR NORMAL_CIRCUIT_CONGESTION > > 2009-12-03 17:21:34.724706 [DEBUG] switch_ivr_originate.c:2988 Originate > Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] > 2009-12-03 17:21:34.725762 [WARNING] mod_openzap.c:474 VETO Changing state > on 1:1 from DOWN to HANGUP > 2009-12-03 17:21:34.725762 [DEBUG] mod_openzap.c:510 OpenZAP/1:1/xxxxxxxx > CHANNEL HANGUP > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:46 > OpenZAP/1:1/xxxxxxxx Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:486 > (OpenZAP/1:1/xxxxxxxx) State HANGUP going to sleep > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:333 > (OpenZAP/1:1/xxxxxxxx) State Change CS_HANGUP -> CS_REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 > (OpenZAP/1:1/xxxxxxxx) State REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:53 > OpenZAP/1:1/xxxxxxxx Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 > (OpenZAP/1:1/xxxxxxxx) State REPORTING going to sleep > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:327 > (OpenZAP/1:1/xxxxxxxx) State Change CS_REPORTING -> CS_DESTROY > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:1136 Session 1 > (OpenZAP/1:1/xxxxxxxx) Locked, Waiting on external entities > 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1154 Session 1 > (OpenZAP/1:1/xxxxxxxx) Ended > 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1156 Close > Channel OpenZAP/1:1/xxxxxxxx [CS_DESTROY] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:423 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_DESTROY > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:434 > (OpenZAP/1:1/xxxxxxxx) State DESTROY > freeswitch at localhost.localdomain> 2009-12-03 17:21:34.726741 [DEBUG] > switch_core_state_machine.c:60 OpenZAP/1:1/xxxxxxxx Standard DESTROY > 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:434 > (OpenZAP/1:1/xxxxxxxx) State DESTROY going to sleep > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c98e19a7/attachment.html From gmaruzz at celliax.org Tue Dec 8 02:14:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 8 Dec 2009 11:14:36 +0100 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> Message-ID: <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > We can ONLY hope someone will do this and BSD/MIT the library and NOT > GPL it... if they GPL it then we'll have to have someone write it all > over again... love the Open Source oil and water. > > /b > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > >>> it I suspect. >> >> Given that they released the codec specification, perhaps someone is >> writing >> an independent C implementation? (Not that I'm much interested, >> but...) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From yehavi.bourvine at gmail.com Tue Dec 8 03:12:37 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Dec 2009 13:12:37 +0200 Subject: [Freeswitch-users] Debugging reeswitch (especially TLS) Message-ID: Hello, I have some black hole understading how to debug Freeswitch. In fs_cli I do "sofia debug all 7" and indeed get a lot of debugging messages on the console; however, the logfiles get only Critical messages. Where do I define which messages go to the logfile? And in a related topic: I've set a Polycom to use TLS with Freeswitch. I see it contacts FS on TCP port 5061, do some exchange, and then quits and does not use TLS. How do I debug TLS from FS side? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/106bb37f/attachment.html From jbr at consiglia.dk Tue Dec 8 03:46:26 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 8 Dec 2009 12:46:26 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not "like": require "luasql.sqlite" env = luasql.sqlite() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) After changing it to require "luasql.sqlite3" env = luasql.sqlite3() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/4764dc8a/attachment-0001.html From codecomplete at free.fr Tue Dec 8 05:43:25 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 05:43:25 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? Message-ID: <26694069.post@talk.nabble.com> Hello I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm getting a software error althought the TDM card seems detected (lspci -v OK). Dahdi was successfully compiled from source code. Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead with OpenZAP? The reason I ask, is that in another forum, someone mentionned "/etc/asterisk/chan_dahdi.conf". Here's the output from dahdi_cfg -vvv: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Here's ls -l /dev/dahdi/: total 0 crw-rw---- 1 root root 196, 254 Dec 8 13:38 channel crw-rw---- 1 root root 196, 0 Dec 8 13:38 ctl crw-rw---- 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw---- 1 root root 196, 253 Dec 8 13:38 timer Has someone succesfully installed Dahdi without Asterisk? Thank you. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694069.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Russell.Mosemann at cune.org Tue Dec 8 06:15:36 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:15:36 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694069.post@talk.nabble.com> Message-ID: <20091208141536.8AD6E421BDE@mail.cune.org> Fred-145 said: > Has someone succesfully installed Dahdi without Asterisk? Of course, and it's working like a charm. DAHDI is a driver. It doesn't care what software uses it. We're using DAHDI with a TE110P PRI T1 card. What is in /proc/dahdi? If it shows "1", what do you see if you "cat /proc/dahdi/1"? Did you correctly configure the files in /etc/dahdi? How did you configure ../freeswitch/conf/openzap.conf? Maybe it would be helpful to spend a few minutes browsing the wiki at http://wiki.freeswitch.org/wiki/OpenZAP -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From kevin at johnnyvoip.com Tue Dec 8 06:17:15 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Tue, 8 Dec 2009 09:17:15 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Message-ID: It seems you can get a copy of either the binaries or the source by doing the following: - Review & execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. - Return executed agreement to silksupport at skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN - Skype will email you the SILK binary once we receive the executed agreement. - Check out documentation, FAQ, and discussion forum (URL TBD) - Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli wrote: > Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... > > On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > > We can ONLY hope someone will do this and BSD/MIT the library and NOT > > GPL it... if they GPL it then we'll have to have someone write it all > > over again... love the Open Source oil and water. > > > > /b > > > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > > > >>> it I suspect. > >> > >> Given that they released the codec specification, perhaps someone is > >> writing > >> an independent C implementation? (Not that I'm much interested, > >> but...) > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/95d9c25b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Copy of Skype - SILK Codec License 27052009.pdf Type: application/pdf Size: 51837 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/95d9c25b/attachment-0001.pdf From mike at jerris.com Tue Dec 8 06:27:24 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 09:27:24 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Message-ID: <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > It seems you can get a copy of either the binaries or the source by > doing the following: > > Review & execute SILK Agreement - attached. NOTE - please add your > Skype login to this form also. > Return executed agreement to silksupport at skype.net and mail hardcopy > to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN > Skype will email you the SILK binary once we receive the executed > agreement. > Check out documentation, FAQ, and discussion forum (URL TBD) > Provide feedback to Skype. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > wrote: > Or it can be LGPL, that's acceptable for FreeSWITCH for my > understanding... > > On Tue, Dec 8, 2009 at 2:50 AM, Brian West > wrote: > > We can ONLY hope someone will do this and BSD/MIT the library and > NOT > > GPL it... if they GPL it then we'll have to have someone write it > all > > over again... love the Open Source oil and water. > > > > /b > > > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > > > >>> it I suspect. > >> > >> Given that they released the codec specification, perhaps someone > is > >> writing > >> an independent C implementation? (Not that I'm much interested, > >> but...) > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/66387092/attachment.html From codecomplete at free.fr Tue Dec 8 06:33:17 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 06:33:17 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <20091208141536.8AD6E421BDE@mail.cune.org> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> Message-ID: <26694801.post@talk.nabble.com> Thanks Russel for the tip. After more googling, I ended up figuring that /etc/dahdi/modules had to contain the list of drivers to load. For those interested, here's how to compile and install Dahdi (which doesn't need Asterisk at all, unlike some docs on the Net seem to imply due to references to /etc/asterisk/*.conf): 1. Download and unpack the Dahdi tarball 2. make all ; make install ; make config 3. cd /etc/dahdi/ 4. vi system.conf: #For France, single FXO module on TDM card loadzone = fr defaultzone = fr fxsks=1 5. vi modules: wcfxo wctdm dahdi 6. /etc/init.d/dahdi start 7. dahdi_cfg -vvv 8. ls -la /proc/dahdi/ Now, on to OpenZAP... Thanks again. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694801.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 8 06:33:55 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 08:33:55 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: > Changing the core db into a MySQL via ODBC caused some problems even > after it seemed to work. For instance, console help caused an error > with an error description indicating that a SQL SELECT query > including the reserved word key has been fired. > > It this problem likely to be solved if I used another version of the > MySQL? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/042c007c/attachment.html From kevin at johnnyvoip.com Tue Dec 8 06:39:22 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Tue, 8 Dec 2009 09:39:22 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > > It seems you can get a copy of either the binaries or the source by doing > the following: > > > - Review & execute SILK Agreement - attached. NOTE - please add your > Skype login to this form also. > - Return executed agreement to > silksupport at skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd > Floor, 2 Stephen Street, London W1T 1AN > - Skype will email you the SILK binary once we receive the executed > agreement. > - Check out documentation, FAQ, and discussion forum (URL TBD) > - Provide feedback to Skype. > > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli < > gmaruzz at celliax.org> wrote: > >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West < >> brian at freeswitch.org> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and NOT >> > GPL it... if they GPL it then we'll have to have someone write it all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > >> FreeSWITCH-users at lists.freeswitch.org >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9566f1ad/attachment-0001.html From Russell.Mosemann at cune.org Tue Dec 8 06:44:40 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:44:40 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> Message-ID: <20091208144440.4092839B743@mail.cune.org> Fred-145 said: > 5. vi modules: > wcfxo > wctdm > dahdi You only need one of the modules above, if you have one card. I don't see a "dahdi" module listed in the file here. > 8. ls -la /proc/dahdi/ You should be able to cat the file in that directory for more information. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From Russell.Mosemann at cune.org Tue Dec 8 06:46:06 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:46:06 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> Message-ID: <20091208144606.18CF03E57BD@mail.cune.org> Fred-145 said: > For those interested, here's how to compile and install Dahdi It would be helpful to others if you add the results of your efforts to the wiki. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From codecomplete at free.fr Tue Dec 8 07:32:42 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 07:32:42 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <20091208144440.4092839B743@mail.cune.org> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> <20091208144440.4092839B743@mail.cune.org> Message-ID: <26695674.post@talk.nabble.com> Russell.Mosemann wrote: > You only need one of the modules above, if you have one card. I don't see > a "dahdi" module listed in the file here. Yup, turns out wcfxo is needed for the X10xP card, while wctdm is needed for Digium cards. As for dahdi, maybe wcfxo/wctdm loads the dahdi module automatically? Russell.Mosemann wrote: > 8. ls -la /proc/dahdi/ You should be able to cat the file in that > directory for more information. Yes indeed: # ls -al /proc/dahdi/ total 0 dr-xr-xr-x 2 root root 0 Dec 8 16:30 . dr-xr-xr-x 80 root root 0 Dec 8 13:37 .. -r--r--r-- 1 root root 0 Dec 8 16:30 1 # cat 1 Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 3 WCTDM/4/2 4 WCTDM/4/3 Thanks for the tip. I'll see if I can update the wiki accordingly. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26695674.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Tue Dec 8 07:49:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Dec 2009 10:49:32 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: They provide you with a 32 bit library, with the header files to link with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 8-Dec-09, at 9:39 AM, Kevin Green wrote: > Their site (https://developer.skype.com/silk) specifies that they > will provide the source, which as you say may not be 64-Bit > compatible but could likely be tweaked to work. I think you just > need to be specific in that you want a source copy not a binary copy > of the codec. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris > wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > >> It seems you can get a copy of either the binaries or the source by >> doing the following: >> >> Review & execute SILK Agreement - attached. NOTE - please add your >> Skype login to this form also. >> Return executed agreement to silksupport at skype.net and mail >> hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, >> London W1T 1AN >> Skype will email you the SILK binary once we receive the executed >> agreement. >> Check out documentation, FAQ, and discussion forum (URL TBD) >> Provide feedback to Skype. >> >> Regards, >> Kevin Green >> >> JohnnyVoIP >> http://www.johnnyvoip.com >> >> >> On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > > wrote: >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West >> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and >> NOT >> > GPL it... if they GPL it then we'll have to have someone write it >> all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps >> someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ad3fe9fd/attachment.html From mike at jerris.com Tue Dec 8 07:58:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 10:58:53 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> We have as of yet been unable to obtain source and we have been in very close contact with skype all the way up to the lead technical and business people on this project. We would of course welcome access to the source but we have as of yet not been able to get a copy Mike On Dec 8, 2009, at 9:39 AM, Kevin Green wrote: > Their site (https://developer.skype.com/silk) specifies that they > will provide the source, which as you say may not be 64-Bit > compatible but could likely be tweaked to work. I think you just > need to be specific in that you want a source copy not a binary copy > of the codec. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris > wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > >> It seems you can get a copy of either the binaries or the source by >> doing the following: >> >> Review & execute SILK Agreement - attached. NOTE - please add your >> Skype login to this form also. >> Return executed agreement to silksupport at skype.net and mail >> hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, >> London W1T 1AN >> Skype will email you the SILK binary once we receive the executed >> agreement. >> Check out documentation, FAQ, and discussion forum (URL TBD) >> Provide feedback to Skype. >> >> Regards, >> Kevin Green >> >> JohnnyVoIP >> http://www.johnnyvoip.com >> >> >> On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > > wrote: >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West >> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and >> NOT >> > GPL it... if they GPL it then we'll have to have someone write it >> all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps >> someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e0fbd696/attachment-0001.html From Prometheus001 at gmx.net Tue Dec 8 08:02:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 08 Dec 2009 17:02:28 +0100 Subject: [Freeswitch-users] Force presence status manually Message-ID: <4B1E7894.2050101@gmx.net> Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. Best regards Peter From spencer at 5ninesolutions.com Mon Dec 7 13:34:05 2009 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 7 Dec 2009 13:34:05 -0800 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers Message-ID: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. From brian at freeswitch.org Tue Dec 8 08:04:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 10:04:32 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: The fun part comes when you try to link that 32bit .a file into a 64bit so file. :P /b On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote: > They provide you with a 32 bit library, with the header files to > link with it. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/220d3e6d/attachment.html From brian at freeswitch.org Tue Dec 8 08:04:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 10:04:44 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> Message-ID: I have resubmitted our request for the source. /b On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote: > We have as of yet been unable to obtain source and we have been in > very close contact with skype all the way up to the lead technical > and business people on this project. We would of course welcome > access to the source but we have as of yet not been able to get a copy > > Mike From mike at jerris.com Tue Dec 8 08:09:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:09:53 -0500 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: <5FA1ADE4-ECF9-4C33-A2FE-2EACB8A946AE@jerris.com> You definitely need to use the settings in combination for what you are trying to do. Can you explain a bit more what you want to do in what conditions and maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure, so it can continue on after the bridge unless you specify otherwise. Mike On Dec 7, 2009, at 1:31 PM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > > data="sofia/external/219 at 10.11.12.243,sofia/external/223 at 10.11.12.234,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245"/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. From mrene_lists at avgs.ca Tue Dec 8 08:11:40 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Dec 2009 11:11:40 -0500 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: set hangup_after_bridge=true Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 7-Dec-09, at 1:31 PM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > data="continue_on_fail=NO_ANSWER,USER_BUSY"/> > data="sofia/external/219 at 10.11.12.243,sofia/external/ > 223 at 10.11.12.234,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245 > "/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not > fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Dec 8 08:12:26 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:12:26 -0500 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> Message-ID: <19B11DE0-0F4A-4641-9B53-D2ED21261D48@jerris.com> If this issue continues after another update and re bootstrap/configure, please open up a bug on jira.freeswitch.org under build system, assign to me, and attach the config.log and config.status file from the root of your freeswitch src dir. Mike On Dec 7, 2009, at 2:39 PM, Anthony Minessale wrote: > try rerunning the ./bootstrap.sh > > > On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards wrote: > When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Monday, December 07, 2009 7:44 AM > To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. > > Best Regards, > Jerry > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Saturday, December 05, 2009 7:30 PM > To: Jerry Richards > Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > >> Why are you changing the 3pcc setting, is this an invite with no sdp? >> you need to take a trace from FS. >> >> 1) update to latest trunk first so line number match up. >> 2) issue these commands >> >> sofia profile internal siptrace on >> console loglevel debug >> >> save the output and put it on pastebin http://pastebin.freeswitch.org >> >> >> >> >> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9c5b118f/attachment-0001.html From mike at jerris.com Tue Dec 8 08:14:29 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:14:29 -0500 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> Please re-test this with svn trunk of freeswitch and if it is still the case open up a bug on jira.freeswitch.org in the build system catagory assigned to me and attach the config.log and config.status from the libs/esl dir to the bug. Mike On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote: > Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Monday, December 07, 2009 1:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 > > The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. > > -SOLINK=-shared -Xlinker -x > +SOLINK=-dynamiclib -Xlinker -x > > > Thats all you usually fix for the mac. > > > /b > > > > On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > > > I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can?t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. > I have also googled this, and don?t see what I am doing wrong. Anybody there that can help? > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > _main > __convert_to_string > __efree > __emalloc > __estrndup > __zend_get_parameters_array_ex > __zend_list_find > __zval_copy_ctor > _compiler_globals > _convert_to_long > _zend_error > _zend_get_constant > _zend_hash_find > _zend_register_list_destructors_ex > _zend_register_long_constant > _zend_register_resource > _zend_rsrc_list_get_rsrc_type > _zend_wrong_param_count > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make: *** [phpmod] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/2e15d6b1/attachment.html From rob4manhere at gmail.com Tue Dec 8 08:14:59 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Tue, 8 Dec 2009 10:14:59 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Message-ID: I'm using FreeSWITCH in front of Asterisk without any issue. Stick with the latest trunk. Can you set your loglevel to debug and pastebin your log? Here are some additional tips to help us help you :) http://wiki.freeswitch.org/wiki/Reporting_Bugs Rob On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/83a2bf78/attachment.html From mike at jerris.com Tue Dec 8 08:16:05 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:16:05 -0500 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > Sorry no, apart from the fact that I was seeing the hangup. > > > I?m wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for ?*? and force a hangup? I don?t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: > Hi all, > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I?m not seeing a hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it?s not being fired. Does anyone have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d0816c10/attachment-0001.html From mike at jerris.com Tue Dec 8 08:19:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:19:42 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> Message-ID: If you can off list provide me with remote login information to this box I can troubleshot the issue. Mike On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote: > Hi Jo?o, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. > > Here's how I normally update FreeSwitch: make clean && svn up && ./bootstrap.sh && ./configure && make install > > If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. > > Regards, > -Jingwei > > > 2009/12/8 Jo?o Mesquita > Maybe, just maybe isse that make target to reconf libtiff? > > Regards, > > JM > > > On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o > at_interpreter.c: In function ???command_search???: > at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use in this function) > at_interpreter.c:5299: error: (Each undeclared identifier is reported only once > at_interpreter.c:5299: error: for each function it appears in.) > at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in this function) > at_interpreter.c: In function ???at_interpreter???: > at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in this function) > make[8]: *** [at_interpreter.lo] Error 1 > > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. > > Regards, > -Jingwei > > > > On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. > > Regards, > -Jingwei > > > On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > Hi, > > That one is on your side. If you changed/updated system libs it might be worth doing another ./configure > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > >> Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. >> >> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc >> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory >> make[8]: *** [at_interpreter_dictionary.h] Error 127 >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> Do you have idea about this one? >> >> Thanks! >> >> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >>> Hi Guys, >>> >>> I got a compilation error of skypiax_protocol.c with the latest version r15764. >>> >>> Compiling skypiax_protocol.c... >>> cc1: warnings being treated as errors >>> skypiax_protocol.c: In function ???X11_errors_handler???: >>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c: In function ???skypiax_send_message???: >>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code >>> make[5]: *** [skypiax_protocol.o] Error 1 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_skypiax-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d424e43e/attachment.html From anthony.minessale at gmail.com Tue Dec 8 08:22:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 10:22:03 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Message-ID: <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> We could check it out for you if you want to contact me and give me ssh access. Or I can provide the instructions get it into the 100% cpu usage state then do the following without stopping FS. 1) run top -H and sort so all the FS threads are at the top and screen cap it so we can see which thread id is using the most cpu. 2) make sure you have gdb installed and issue this command from the build root ./support-d/fscore_pb gcore cpu_race_issue then we can compare the thread using the most cpu with the trace and locate your problem. On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ede6c2b8/attachment.html From anthony.minessale at gmail.com Tue Dec 8 08:28:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 10:28:04 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel wrote: > I got the combination Lua with direct access to the core Sqlite database > to work. Hurray, maybe I?m not as stupid as A.M II hints? > > The problem was that Lua did not ?like?: > > > > require "luasql.sqlite" > > env = luasql.sqlite() > > con = assert(env:connect("/usr/local/freeswitch/db/core.db")) > > > > After changing it to > > > > require "luasql.sqlite3" > > env = luasql.sqlite3() > > con = assert(env:connect("/usr/local/freeswitch/db/core.db")) > > > > And seeing that there was a symlink in one of the right directories called > with the name: sqlite3.so, it worked. > > > > Changing the core db into a MySQL via ODBC caused some problems even after > it seemed to work. For instance, console help caused an error with an error > description indicating that a SQL SELECT query including the reserved word > key has been fired. > > > > It this problem likely to be solved if I used another version of the MySQL? > > > > *Jon Br?el* > Consiglia Telecommunications > > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > > CVR: 27047882 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/74c4d067/attachment-0001.html From mike at jerris.com Tue Dec 8 08:40:40 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:40:40 -0500 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> References: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> Message-ID: <3D5FFA01-DCA1-48E4-970C-8635B2F5F50E@jerris.com> I changed the name of key to ikey in trunk. Mike > Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. > > > It this problem likely to be solved if I used another version of the MySQL? > > > Jon Br?el > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ca6d85a1/attachment.html From mike at jerris.com Tue Dec 8 08:42:38 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:42:38 -0500 Subject: [Freeswitch-users] Force presence status manually In-Reply-To: <4B1E7894.2050101@gmx.net> References: <4B1E7894.2050101@gmx.net> Message-ID: The best way to solve this is probably to share the db for presence and registration between those boxes. If you take a look at the default configs the settings should be commented there. Mike On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote: > Hello, > > is there a way to manually force a presence status update? > In our scenario we have a Freeswitch cluster. As phones sometimes > register on one and one time on another machine via the load balancer, > we cannot dial via user/exten. Instead we dial each phone by it's > register string via xml-curl. That way -when a phone is called - other > phones who subscribed to this phone, do not receive a message to update > their presence status. > Is there a way to force the pesence status of a phone manually in the > dialplan? > We may then set the status before bridging and then reset it with a > hangup hook. > From msc at freeswitch.org Tue Dec 8 09:39:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Dec 2009 09:39:16 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <87f2f3b90912080939s473feda4n7ea21fc995905b25@mail.gmail.com> On Tue, Dec 8, 2009 at 3:46 AM, Jon Bruel wrote: > I got the combination Lua with direct access to the core Sqlite database > to work. Hurray, maybe I?m not as stupid as A.M II hints? > Tsk tsk! He didn't actually hint that you were "stupid" - all he said was that doing ODBC and configuring databases isn't something as simple as flipping on a switch. It takes a bit of knowledge, much of which is hard-earned through experience. Trying to get it all up and running by emailing the list every time something goes wrong is like trying to learning how to change the oil in your car and emailing the Audi-users list every time something doesn't go as expected: yeah, you can probably learn something, possibly you can get it working, but it's grossly inefficient. You'd be much better off paying someone a few euros to come out and give you a lesson because in the long run it would save you both time and money. Just my $0.02. (Don't know what that is in euros...) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c3c40e9c/attachment.html From abeka at greatiam.com Tue Dec 8 09:47:53 2009 From: abeka at greatiam.com (Otis) Date: Tue, 08 Dec 2009 17:47:53 +0000 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <4B1D4E92.1040204@greatiam.com> References: <4B1CAF25.6010706@greatiam.com> <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> <4B1D4E92.1040204@greatiam.com> Message-ID: <4B1E9149.80302@greatiam.com> Otis wrote: >
Michael > Collins wrote: >> >> >> On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah >> > wrote: >> >> Pardon me if this has been addressed already. >> How does one go about having in the simplest instance 2 servers >> registering with each other on startup whereby the users registering >> would be able to call each other. >> The 2 servers are in different domains. >> >> Thanks. >> >> >> Are the two servers in different locations? Different LANs? Is NAT >> involved? Just checking. Really this is just a matter of loading the >> default config on each machine and then making some decisions about >> the dialplan: do you want prefix dialing so that you can have ext >> 1000 at both locations or do you want to have something like >> 1000~1099 at location A and 1100~1199 at location B? From there it's >> just a matter of creating the gateways on each machine and adding a >> dialplan entry to handle the routing. >> -MC >> > Hello Michael > Thanks > Are the two servers in different locations? Yes > Different LANs? Yes > Is NAT involved? Yes but for my test Nat is not . The production setup > I have in mind will certainly have Nat > Each location will have their won set of extension but there could be > some overlap. > On server A a user would dial,. for example, 98 followed by the > extension number of the user on server B and the call would then be > routed to the extension on server B. And the same could be from > Server B to a user on Server A > > MC > > Thanks > > . > > >
> Please olks could someone let meknow if it is at possible. I have tried using the connecting to Asterisk without success, mimicked the link to a gateway unsuccessfully. Could someone please let me kno which .xml files to create etc. Thanks From Russell.Mosemann at cune.org Tue Dec 8 08:38:07 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 16:38:07 -0000 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: Message-ID: <20091208163807.DC92C45048A@mail.cune.org> Brian West said: > The fun part comes when you try to link that 32bit .a file into a > 64bit so file. That would require a dual-core processor. One core would be 32 bit and the other core would be 64 bit. ;-) -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From brian at freeswitch.org Tue Dec 8 10:00:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 12:00:05 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <20091208163807.DC92C45048A@mail.cune.org> References: <20091208163807.DC92C45048A@mail.cune.org> Message-ID: Well the fun part is you can't link them. :P /b On Dec 8, 2009, at 10:38 AM, wrote: > That would require a dual-core processor. One core would be 32 bit and > the other core would be 64 bit. ;-) > > -- > Russell Mosemann From jbr at consiglia.dk Tue Dec 8 10:00:54 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 8 Dec 2009 19:00:54 +0100 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> References: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> Message-ID: Point taken Anthony. Naturally you are not going to work for me for free. But I'm a bit confused about the statement that "I'm rude". That's not my purpose to be. And I certainly do hope that this is not just a question of a cultural clash between an elderly man with a Phd in black holes from a European background and a young American FS genius. But frankly, I did believe that focus regarding changes and new developments was somewhat guided by the input you get from the users list, including changes which makes the FS easier to access for newbies, but maybe I'm wrong. That's my last comment, hope we can continue the exchange of views in a good spirit. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 8. december 2009 17:28 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel > wrote: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not "like": require "luasql.sqlite" env = luasql.sqlite() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) After changing it to require "luasql.sqlite3" env = luasql.sqlite3() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/7b83ab6d/attachment-0001.html From xengelpublicx at gmail.com Tue Dec 8 10:14:02 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Tue, 08 Dec 2009 21:14:02 +0300 Subject: [Freeswitch-users] presense. freeswitch vs. spa962+932 Message-ID: <4B1E976A.5080207@gmail.com> Hello, I tune in the presence freeswitch and linksys spa962 932 I had a few qusteions: 1) if $ PROXY specified domain name and not ip the phone records. But all the buttons on spa932 blinking orange indicating that no subscriptions. phone logs like this: Call-ID: 76e0f816-9617ab46 at 192.168.0.100 User: 100 at 192.168.50.10 Contact: "user" Agent: Linksys/SPA962-6.1.3 (a) Status: Registered (UDP-NAT) (unknown) EXP (2009-12-08 21:26:14) Host: pbx0.test.lan IP: 192.168.0.100 Port: 1024 Auth-User: 100 Auth-Realm: pbx0.test.lan MWI-Account: 100 at 192.168.50.10 while the phone is not a nat. spa932 shows that subscriptions present. 2) how to see that now there is a basis of presence of fs_cli? 3) Can I configure two fs a mutually shared presence? This is done using ? Thabks. From spencer at 5ninesolutions.com Tue Dec 8 10:14:42 2009 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 8 Dec 2009 10:14:42 -0800 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> Message-ID: Hmm.. It doesn't seem to be a problem with Asterisk < 1.6.0.13. Asterisk 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this after an upgrade to 1.6.0.19. We're using xen on all our machines with 250hz timers. Could that be a problem? When I get a change I'll try to recreate this with a few more virtual machines to try to debug it. Spencer On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote: > We could check it out for you if you want to contact me and give me > ssh access. > Or I can provide the instructions > > get it into the 100% cpu usage state then do the following without > stopping FS. > > 1) run top -H and sort so all the FS threads are at the top and > screen cap it so we can see which thread id is using the most cpu. > 2) make sure you have gdb installed and issue this command from the > build root > ./support-d/fscore_pb gcore cpu_race_issue > > then we can compare the thread using the most cpu with the trace and > locate your problem. > > > > On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason > wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the > cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/b127e2ae/attachment.html From msc at freeswitch.org Tue Dec 8 10:19:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Dec 2009 10:19:39 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here! Message-ID: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> Greetings, The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH 1.0.5 pre-release version. Please check out the release announcement. Let's all get updated as soon as possible. Also, please report bugs right away and follow up when the developers need further information. We have had to close out some bugs due to lack of information from the one reporting. Of course, those running SVN trunk are asked to do a "make current" as soon as reasonably possible. The devs love it when you are on the latest trunk. :) Thanks again for all of your help! Let's keep up the good work and we'll have 1.0.5 available in no time. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/1e42be41/attachment.html From anthony.minessale at gmail.com Tue Dec 8 10:23:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 12:23:39 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> Message-ID: <191c3a030912081023tdbdf0efjfd474b53214ca930@mail.gmail.com> would not be able to even guess without some data to examine. On Tue, Dec 8, 2009 at 12:14 PM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > Hmm.. It doesn't seem to be a problem with Asterisk < 1.6.0.13. Asterisk > 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this > after an upgrade to 1.6.0.19. We're using xen on all our machines with > 250hz timers. Could that be a problem? When I get a change I'll try to > recreate this with a few more virtual machines to try to debug it. > > Spencer > > On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote: > > We could check it out for you if you want to contact me and give me ssh > access. > Or I can provide the instructions > > get it into the 100% cpu usage state then do the following without stopping > FS. > > 1) run top -H and sort so all the FS threads are at the top and screen cap > it so we can see which thread id is using the most cpu. > 2) make sure you have gdb installed and issue this command from the build > root > ./support-d/fscore_pb gcore cpu_race_issue > > then we can compare the thread using the most cpu with the trace and locate > your problem. > > > > On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason < > spencer at 5ninesolutions.com> wrote: > >> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. >> The Asterisk boxes are individual hosted PBXs but they are configured >> with identical software. This a x86_64 CentOS 5.4 system. I've tried >> 1.0.4 and the latest svn with the same results. Basically Freeswitch >> registers with outbound providers and I can send and receive test >> calls. Then without warning, i.e. the Asterisk boxes are all idle and >> there are no calls, the Freeswitch process starts using 100% of the cpu. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e2e659f6/attachment.html From anthony.minessale at gmail.com Tue Dec 8 10:25:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 12:25:52 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here! In-Reply-To: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> References: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> Message-ID: <191c3a030912081025l1481cda9rc7cbbd0343ef51cc@mail.gmail.com> Let's see if we can beat Duke Nukem Forever! On Tue, Dec 8, 2009 at 12:19 PM, Michael Collins wrote: > Greetings, > > The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH > 1.0.5 pre-release version. Please check out the release announcement. > Let's all get updated as soon as possible. Also, please report bugs right > away and follow up when the developers need further information. We have had > to close out some bugs due to lack of information from the one reporting. > > Of course, those running SVN trunk are asked to do a "make current" as soon > as reasonably possible. The devs love it when you are on the latest trunk. > :) > > Thanks again for all of your help! Let's keep up the good work and we'll > have 1.0.5 available in no time. > > -Michael > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f267b1bf/attachment-0001.html From JCasale at activenetwerx.com Tue Dec 8 10:26:58 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 8 Dec 2009 18:26:58 +0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> Message-ID: >For those interested, here's how to compile and install Dahdi (which doesn't >need Asterisk at all, unlike some docs on the Net seem to imply due to >references to /etc/asterisk/*.conf): I understand that Some Debian based distro's have Dahdi in their repo's making it simple, but not many know that Digium runs its own repo for rpm based distros: http://packages.asterisk.org/ Can't get easier than that... jlc From kristian.kielhofner at gmail.com Tue Dec 8 10:30:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 8 Dec 2009 13:30:12 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <2d9149cd0912081030n14729f1en1ff0ec1a1506357@mail.gmail.com> For reference, here is the AstLinux kernel config for the ALIX: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/target/device/alix/linux.config?view=markup We've got what I consider to be excellent support for the ALIX - most of the developers use them and they are very popular in the community. On Mon, Dec 7, 2009 at 11:00 AM, Anthony Minessale wrote: > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm.? We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using.? I am curious > what would happen if you install Kristian's astlinux on one of your devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do these > systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread > which will make all the switch_cond_next share a 1ms conditional instead of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others work > better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From testa at voicetechnology.com.br Tue Dec 8 10:41:15 2009 From: testa at voicetechnology.com.br (Fernando Gregianin Testa) Date: Tue, 8 Dec 2009 16:41:15 -0200 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones Message-ID: Dear list, Some Nec phones sends DTMF RFC2833 with payload 101 during the call, but have negotiated a different one on SDP. When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 we notice this phone sends the following INVITE packet and RTP packets: http://pastebin.freeswitch.org/11433 Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap Is there any parameter to tweak FS in such a way to force understand 101 packets as DTMF? Thank you in advance! Fernando Testa PS: On pcap you have the following IPs: FS at 10.91.10.210 Nec Pbx 10.91.10.22 phone 10.91.10.85 From brian at freeswitch.org Tue Dec 8 10:51:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 12:51:32 -0600 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones In-Reply-To: References: Message-ID: Best option for you is to use 96 in the sofia profile you're using to talk to these broken devices. /b On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: > Dear list, > > Some Nec phones sends DTMF RFC2833 with payload 101 during the call, > but have negotiated a different one on SDP. > When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 > we notice this phone sends the following INVITE packet and RTP > packets: http://pastebin.freeswitch.org/11433 > Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap > > Is there any parameter to tweak FS in such a way to force understand > 101 packets as DTMF? > Thank you in advance! > > Fernando Testa > PS: On pcap you have the following IPs: > FS at 10.91.10.210 > Nec Pbx 10.91.10.22 > phone 10.91.10.85 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From ken at ksac.com Tue Dec 8 10:56:36 2009 From: ken at ksac.com (Kendall Stauffer) Date: Tue, 8 Dec 2009 10:56:36 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> References: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> Message-ID: Hey you guys, I know this isn't the right place for this, but I have been working with asterisk for 5 years now, and just got freeswitch working (on windows, not os x yet). All I can say is AWESOME --- thanks so much!!!! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 08, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d3af9e8f/attachment.html From mailinglist at fribert.dk Tue Dec 8 11:20:29 2009 From: mailinglist at fribert.dk (mailinglist) Date: Tue, 08 Dec 2009 20:20:29 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1DFABC020000E1000002C2@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> Message-ID: <4B1EB50D020000E1000002C7@mail.fribert.dk> Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f925530e/attachment-0001.html From mike at jerris.com Tue Dec 8 11:57:28 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 14:57:28 -0500 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> Message-ID: <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> Our plan for 1.0.5 is that we will also have rpm and deb packages for many distros on our own repo. Stay tuned. This has been another major reason for the delay in 1.0.5. Mike On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote: >> For those interested, here's how to compile and install Dahdi (which doesn't >> need Asterisk at all, unlike some docs on the Net seem to imply due to >> references to /etc/asterisk/*.conf): > > I understand that Some Debian based distro's have Dahdi in their repo's making it > simple, but not many know that Digium runs its own repo for rpm based distros: > > http://packages.asterisk.org/ > > Can't get easier than that... > > jlc > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jerry.richards at teotech.com Tue Dec 8 12:35:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 Dec 2009 12:35:01 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/81bdc073/attachment.html From jerry.richards at teotech.com Tue Dec 8 12:57:39 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 Dec 2009 12:57:39 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9755354a/attachment-0001.html From larclap at yahoo.com Tue Dec 8 13:09:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 8 Dec 2009 13:09:03 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> Message-ID: <011401ca784a$ac065690$041303b0$@com> Can you copy the address of the pastebin so that people can see it? After you hit the Send button, the address is posted back at the top of your browser, like: http://pastebin.freeswitch.org/11441 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c7eef752/attachment.html From anthony.minessale at gmail.com Tue Dec 8 13:21:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 15:21:14 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <191c3a030912081321j5c875e67r5a0ef6f6d018fa65@mail.gmail.com> are you using more than one profile here? if so you have to repeat the siptrace on command for each one. This trace makes little sense to me because I think half of it is missing. but you can see several packets coming in like 20 times each which means you have some kind of nat or network problem causing the other end of this call to send retries on all the packets. On Tue, Dec 8, 2009 at 2:57 PM, Jerry Richards wrote: > Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 > > Thanks, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Tuesday, December 08, 2009 12:35 PM > > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Anthony and Michael, > > I downloaded the latest trunk, rebuilt it, and re-ran the test with the > logs that Anthony told me to turn on. I put the results up in the PasteBin. > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 10:49 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > When I got the latest trunk the make gets an error. Should I perhaps > disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. > Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 7:44 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs > without SDP. I will try to update to the latest trunk today and capture > traces as Anthony described. If I can't do it today, it might be at the end > of the week. > > Best Regards, > Jerry > > > ------------------------------ > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Saturday, December 05, 2009 7:30 PM > *To:* Jerry Richards > *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > > Why are you changing the 3pcc setting, is this an invite with no sdp? > you need to take a trace from FS. > > 1) update to latest trunk first so line number match up. > 2) issue these commands > > sofia profile internal siptrace on > console loglevel debug > > save the output and put it on pastebin http://pastebin.freeswitch.org > > > > > On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards > wrote: > >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/660ad668/attachment-0001.html From orien at tx.rr.com Tue Dec 8 13:25:22 2009 From: orien at tx.rr.com (Orien Love) Date: Tue, 08 Dec 2009 15:25:22 -0600 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: References: Message-ID: <4B1EC442.7010603@tx.rr.com> I am looking for a 4 port FXO card to use with my PfSense installation of freeswitch. does anybody know if the Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? or could somebody recommend one that would. Thank You Orien From anthony.minessale at gmail.com Tue Dec 8 13:33:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 15:33:30 -0600 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: <4B1EC442.7010603@tx.rr.com> References: <4B1EC442.7010603@tx.rr.com> Message-ID: <191c3a030912081333t23c27bcbh720fa5930a413ff@mail.gmail.com> I dont think there are any supported hw for bsd, there are legacy sangoma and zaptel drivers floating around but they are not supported by the vendors. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love wrote: > I am looking for a 4 port FXO card to use with my PfSense installation > of freeswitch. does anybody know if the > Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > or could somebody recommend one that would. > > Thank You > Orien > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/1242b553/attachment.html From rsavage at KingBallow.com Tue Dec 8 13:36:23 2009 From: rsavage at KingBallow.com (Reece Savage) Date: Tue, 8 Dec 2009 15:36:23 -0600 Subject: [Freeswitch-users] Rhino Cards for sale R2T1-EC and R24FXX-EC In-Reply-To: References: <26694069.post@talk.nabble.com><20091208141536.8AD6E421BDE@mail.cune.org><26694801.post@talk.nabble.com> Message-ID: <8E4ACA7747F7F641991455BC157390C801450356@srv-nash-ex.mail.kingballow.com> I have 2 Rhino cards for sale if anyone needs one. They are both Best Offer. I have a R2T1-EC and a R24FXX-EC with 12 dual FXS modules. Both have never been used more than a few times for testing purposes. Both cards work fine and are guaranteed not to be DOA. Reece Savage Information Technology Manager King & Ballow Law Offices 315 Union Street Suite 1100 Nashville, TN? 37201 Phone (615) 726-5525 Fax (615) 254-7907 rsavage at kingballow.com From nandy1925 at gmail.com Tue Dec 8 14:45:15 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 9 Dec 2009 06:45:15 +0800 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1EB50D020000E1000002C7@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> Message-ID: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: > Hi All > > Ok, after reading a bit more I think I see what I've done wrong, but I > don't know how to fix it properly. > Looking in the Dialplan directory I see the following: > default (dir) > default.xml > features.xml > public (dir) > public.xml > > Under the default dir the webinterface has created the 001_musimi.dk.xml > file that I've created. > But as I understand it, it doesn't use it. > > How do I make it use it, I would very much like to keep the webinterface > editor, and not have to do it via ssh and vi all the time. > > >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i > meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: > Hi Mark > > Ok, thanks. > Yes I have a gateway placed in external called musimi.dk (or should it be > in public?), and I'll just create the empty XML's in lan to get rid of that > error. > > I'll remove the second part of the dialplan, my idea was that it was needed > for calls between sip phones hooked up to the freeswitch. > > Now the remaining problem: > When I call ext 1002 from ext 1001 I see this message and get an error, the > same goes for dialing 0XXXXXXXX to get an external number: > > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing > 1001->1002 in context default > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 > [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. > Cause: NO_ROUTE_DESTINATION > 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup > sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 > (sofia/external/$1) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/external/$1 [CS_DESTROY] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( > sofia/internal/1001 at 10.11.12.25) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] > I don't see any mention of the statements in the Dialplan, so for me it > looks like it haven't registered the Dialplan? > > Best regards > Kenneth > > >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen > 659603.29094.qm at web56408.mail.re3.yahoo.com: > > Question ---------------------------------------------- > If I do a reloadxml it gives me this output on the console: > freeswitch at firewall.fribert.dk> > reloadxml > 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > > I'm not sure if it's a genuine problem,as I can see it, it just complains > that I haven't created any sip_profiles in /lan, but is that necessary? > > Response: ---------------------------------------------- > This isn't really a problem. To get rid of the error simply put a blank xml > file into each folder as in the internal and external directories. Dump the > lan directory and lan profile as mentioned earlier. > > Question ---------------------------------------------- > > Extension Name musimi.dk > Enabled true > Order 001 > Description ... > > condition ^0(.\d+)$ > action bridge sofia/gateway/musimi.dk/$1 > > Response: ---------------------------------------------- > > This is correct as long as you have a gateway that is registered called > musimi.dk > > Question ---------------------------------------------- > Extension Name 10.11.12.25 > Enabled true > Order 002 > Description ... > > action bridge sofia/internal/$ > > Response: ---------------------------------------------- > > No idea what this is for its not needed as far as I can tell. > > > Now please summarize what you still need help on. > > > Mark J Crane > http://fusionpbx.com > pfSense FreeSWITCH package developer > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/b0204313/attachment.html From nandy1925 at gmail.com Tue Dec 8 14:50:58 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 9 Dec 2009 06:50:58 +0800 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> this action can be accomplished using Group Dialing (Sequential). this may not answer your problem but have you considered it? -nandy On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > > data="sofia/external/219 at 10.11.12.243,sofia/external/223 at 10.11.12.234 > ,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245"/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/9a7a9515/attachment-0001.html From timuckun at gmail.com Tue Dec 8 16:24:11 2009 From: timuckun at gmail.com (Tim Uckun) Date: Wed, 9 Dec 2009 13:24:11 +1300 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> Message-ID: <855e4dcf0912081624g114ccb47hb7bcfca957c4cc38@mail.gmail.com> On Wed, Dec 9, 2009 at 7:56 AM, Kendall Stauffer wrote: > Hey you guys, I know this isn?t the right place for this, but I have been > working with asterisk for 5 years now, and just got freeswitch working (on > windows, not os x yet). > > All I can say is AWESOME --- thanks so much!!!! Out of curiosity. Did you choose freeswitch because it runs on windows and asterisk doesn't? I find some people choose freeswitch because they don't know or want to use linux (obviously this doesn't apply to you) and some people choose it because they want a windows solution and asterisk doesn't run on windows. From ryannyl at gmail.com Tue Dec 8 19:21:13 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 9 Dec 2009 11:21:13 +0800 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. Message-ID: <4bfcac7e0912081921m1f26d337we9e92b310ea2cd1e@mail.gmail.com> hi I don't know this driver for freebsd works or not, maybe you can check it out and try it. Here is the Digium's SVN repository: http://svn.digium.com/svn/dahdi/ It looks like driver for freebsd is under testing. Or use zaptel driver ... Change to root, then cd /usr/ports/misc/zaptel make install && make clean uh..., maybe PfSense doesn't have /usr/ports ...you can just install zaptel. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love > wrote: > > >* I am looking for a 4 port FXO card to use with my PfSense installation > *>* of freeswitch. does anybody know if the > *>* Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > *>* or could somebody recommend one that would. > *>* > *>* Thank You > *>* Orien > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > *>* http://www.freeswitch.org > *>** > > -- Sincerely regards, Wen-Jen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/2cb28b50/attachment.html From mctch at yahoo.com Wed Dec 9 01:26:49 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 01:26:49 -0800 (PST) Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: <4bfcac7e0912081921m1f26d337we9e92b310ea2cd1e@mail.gmail.com> Message-ID: <446840.56954.qm@web56405.mail.re3.yahoo.com> You need to use a zaptel package that was compiled for the operating system that pfsense uses. For pfSense 1.2.3 try this package:? ssh to pfsense then run: press 8 for the command line access cd /tmp fetch http://portableusbapps.com/packages/config/freeswitch/zaptel-1.4.6_7.tgz pkg_add zaptel-1.4.6_7.tgz Also you will need to use the FreeSWITCH Dev package on pfsense which has the openzap module. FreeSWITCH dev package is not perfect that is why it is still marked as dev. Best Regards, Mark J Crane P.S. For those that don't know the I created the pfSense FreeSWITCH package wanted a name for the project and finally after a lot of thought came up with FusionPBX. Last night I stayed up all night to see if I could get FusionPBX to work on pfSense. I did get it so that is a step in the right direction to get the latest version working on pfSense again. FusionPBX uses PDO (PHP Data Objects) which is available in PHP5. pfSense 1.2.3 uses PHP4 and so it required getting both PHP4 and PHP5 working on the same machine setup in a way that they don't conflict which I was able to achieve. So new version is getting closer. --- On Tue, 12/8/09, Ryanny Lin wrote: From: Ryanny Lin Subject: Re: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 8:21 PM hi I don't know this driver for freebsd works or not, maybe you can check it out and try it. Here is the Digium's SVN repository: http://svn.digium.com/svn/dahdi/ It looks like driver for freebsd is under testing. Or use zaptel driver ... Change to root, then cd /usr/ports/misc/zaptel make install && make clean uh..., maybe PfSense doesn't have /usr/ports ...you can just install zaptel. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love wrote: > I am looking for a 4 port FXO card to use with my PfSense installation > of freeswitch. does anybody know if the > Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > or could somebody recommend one that would. > > Thank You > Orien > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely regards, Wen-Jen -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/ae94c7ba/attachment.html From mctch at yahoo.com Wed Dec 9 01:28:07 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 01:28:07 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <187489.95329.qm@web56408.mail.re3.yahoo.com> Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch at yahoo.com --- On Tue, 12/8/09, Nandy Dagondon wrote: From: Nandy Dagondon Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All ? Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml ? Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark ? Ok, thanks. Yes I have a gateway placed in external?called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan?to get rid of that error. ? I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. ? Now the remaining problem: When I call ext 1002 from?ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: ? 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? ? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console:freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/3eaa110b/attachment-0001.html From Prometheus001 at gmx.net Wed Dec 9 02:12:04 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 09 Dec 2009 11:12:04 +0100 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> References: <4B1D4A08.80507@gmx.net> <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> Message-ID: <4B1F77F4.6030302@gmx.net> Hello Nandy, thanks for your hint, but it's a bit more than that. In our application which is handled via XML-Curl, the user can define it's forwards on a web interface. He can enter mixed local or external numbers which are called sequentially or in parallel. Best regards Peter Nandy Dagondon schrieb: > this action can be accomplished using Group Dialing (Sequential). this > may not answer your problem but have you considered it? > -nandy > > > On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX > wrote: > > I have a Problem with continue_on_fail. > > > I have setup a hunt group > data="continue_on_fail=NO_ANSWER,USER_BUSY"/> > data="sofia/external/219 at 10.11.12.243 > ,sofia/external/223 at 10.11.12.234 > ,sofia/external/236 at 10.11.12.188 > ,sofia/external/101 at 10.11.12.245 > "/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would > not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jonas.gauffin at gmail.com Wed Dec 9 02:25:36 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 9 Dec 2009 11:25:36 +0100 Subject: [Freeswitch-users] OT: Spa2102 and call transfer Message-ID: Hello, I can't get call transfer to work with a SPA2102 adapter. I don't think it has something to do with FS, but I'm hoping someone here could help me. I do not get a new line in the phone (by pressing the R button), all DTMF tones are sent as audio to the other connected phone. Anyone got it working? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/60f8b949/attachment.html From timuckun at gmail.com Wed Dec 9 02:26:41 2009 From: timuckun at gmail.com (Tim Uckun) Date: Wed, 9 Dec 2009 23:26:41 +1300 Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <987536.45831.qm@web37508.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> Message-ID: <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> On Tue, Dec 8, 2009 at 5:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. ?Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > ?Any suggestion would be appreciated. ?Thank you. If you could explain how you are doing the load balancing it would be really helpful to me. I am trying to do the same thing. From devel at thom.fr.eu.org Wed Dec 9 03:01:22 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 09 Dec 2009 12:01:22 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Message-ID: <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> I'm still working on this issue, and decided to take a look at the openzap code. First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid. I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span) By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it. Is my asumption (feature not implemented) correct ? Fran?ois On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitch.org [9] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [10] ClueCon http://www.cluecon.com/ [11] Twitter: http://twitter.com/FreeSWITCH_wire [12] AIM: anthm MSN:anthony_minessale at hotmail.com [13] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [14] IRC: irc.freenode.net [15] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [16] iax:guest at conference.freeswitch.org/888 [17] googletalk:conf+888 at conference.freeswitch.org [18] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [23] ClueCon http://www.cluecon.com/ [24] Twitter: http://twitter.com/FreeSWITCH_wire [25] AIM: anthm MSN:anthony_minessale at hotmail.com [26] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [27] IRC: irc.freenode.net [28] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [29] iax:guest at conference.freeswitch.org/888 [30] googletalk:conf+888 at conference.freeswitch.org [31] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [32] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [33] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [34] http://www.freeswitch.org [35] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [36] ClueCon http://www.cluecon.com/ [37] Twitter: http://twitter.com/FreeSWITCH_wire [38] AIM: anthm MSN:anthony_minessale at hotmail.com [39] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [40] IRC: irc.freenode.net [41] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [42] iax:guest at conference.freeswitch.org/888 [43] googletalk:conf+888 at conference.freeswitch.org [44] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:anthony.minessale at gmail.com [3] mailto:devel at thom.fr.eu.org [4] http://files.freeswitch.org [5] mailto:devel at thom.fr.eu.org [6] mailto:FreeSWITCH-users at lists.freeswitch.org [7] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] http://lists.freeswitch.org/mailman/options/freeswitch-users [9] http://www.freeswitch.org [10] http://www.freeswitch.org/ [11] http://www.cluecon.com/ [12] http://twitter.com/FreeSWITCH_wire [13] mailto:MSN%3Aanthony_minessale at hotmail.com [14] mailto:PAYPAL%3Aanthony.minessale at gmail.com [15] http://irc.freenode.net [16] mailto:sip%3A888 at conference.freeswitch.org [17] http://iax:guest at conference.freeswitch.org/888 [18] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [19] mailto:FreeSWITCH-users at lists.freeswitch.org [20] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21] http://lists.freeswitch.org/mailman/options/freeswitch-users [22] http://www.freeswitch.org [23] http://www.freeswitch.org/ [24] http://www.cluecon.com/ [25] http://twitter.com/FreeSWITCH_wire [26] mailto:MSN%3Aanthony_minessale at hotmail.com [27] mailto:PAYPAL%3Aanthony.minessale at gmail.com [28] http://irc.freenode.net [29] mailto:sip%3A888 at conference.freeswitch.org [30] http://iax:guest at conference.freeswitch.org/888 [31] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [32] mailto:FreeSWITCH-users at lists.freeswitch.org [33] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] http://lists.freeswitch.org/mailman/options/freeswitch-users [35] http://www.freeswitch.org [36] http://www.freeswitch.org/ [37] http://www.cluecon.com/ [38] http://twitter.com/FreeSWITCH_wire [39] mailto:MSN%3Aanthony_minessale at hotmail.com [40] mailto:PAYPAL%3Aanthony.minessale at gmail.com [41] http://irc.freenode.net [42] mailto:sip%3A888 at conference.freeswitch.org [43] http://iax:guest at conference.freeswitch.org/888 [44] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/c963a88f/attachment.html From nik.middleton at noblesolutions.co.uk Tue Dec 8 15:59:55 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 8 Dec 2009 23:59:55 -0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! Message-ID: Thought I'd send this little hurrah! As there seems to have been a lot of negativity on this list lately. >From my point of view, having looked at many solutions out there, FS is still number one with regards to flexibility and performance. I cannot imagine doing what I'm using FS for, with any other product. Yes it's frustrating at times, but this is largely down to a lack documentation/samples. So, if you have a solution to a problem, share it by adding an entry on the WIKI. Kudos to AM and all the other dev's, as someone said once 'Don't let the bastards grind you down' Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f9b84f4e/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Dec 8 15:48:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 8 Dec 2009 23:48:58 -0000 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: No doubt, but that's a little difficult as this only happens occasionally and I have 200 calls going on at the time. It's needle in the haystack stuff. Here's what I know. I have an external process listening for DTMF events. If I detect '*' I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn't exist, so I now do a double kill on the associated leg which I get from the event. I do not get a 'doesn't exist' message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. The kluge clears the B leg hang issue, so the pressure's off for me, but when I get a few nano seconds, I'll look at the code to see if there's anything obvious. Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 08 December 2009 16:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/21743be3/attachment.html From codecomplete at free.fr Wed Dec 9 03:34:24 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 9 Dec 2009 03:34:24 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> Message-ID: <26708848.post@talk.nabble.com> Michael Jerris wrote: > Our plan for 1.0.5 is that we will also have rpm and deb packages for many > distros on our own repo. Stay tuned. This has been another major reason > for the delay in 1.0.5. Great news. I also prefer to use packages whenever possible, so as to know what software is installed in a host, and have the package manager handle conflicts and missing dependencies. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26708848.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Dec 9 05:58:46 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Dec 2009 08:58:46 -0500 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> Message-ID: <76C4CA2A-4569-4EAA-83CF-E0EEDFC18242@jerris.com> I recall implementing that back when we released openzap, it should be in there unless someone chopped it out for some reason. Look for "zap_channel_send_fsk_data" Mike On Dec 9, 2009, at 6:01 AM, Fran?ois Legal wrote: > I'm still working on this issue, and decided to take a look at the openzap code. > > First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid. > > I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span) > > > By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it. > > > Is my asumption (feature not implemented) correct ? > > > Fran?ois > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/2cf60bed/attachment.html From mike at jerris.com Wed Dec 9 06:00:43 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Dec 2009 09:00:43 -0500 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> src/switch_ivr_bridge.c This could just as well be a glare condition when the call is in process of tearing down. Mike On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote: > No doubt, but that?s a little difficult as this only happens occasionally and I have 200 calls going on at the time. It?s needle in the haystack stuff. > > Here?s what I know. > > I have an external process listening for DTMF events. If I detect ?*? I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn?t exist, so I now do a double kill on the associated leg which I get from the event. I do not get a ?doesn?t exist? message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. > > The kluge clears the B leg hang issue, so the pressure?s off for me, but when I get a few nano seconds, I?ll look at the code to see if there?s anything obvious. > > Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: 08 December 2009 16:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > We will really need debug logs and sip traces to be able to figure out what exactly is going on here. > > Mike > > On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > > > Sorry no, apart from the fact that I was seeing the hangup. > > > I?m wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for ?*? and force a hangup? I don?t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: > Hi all, > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I?m not seeing a hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it?s not being fired. Does anyone have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/8260f359/attachment-0001.html From jonas.gauffin at gmail.com Wed Dec 9 06:01:14 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 9 Dec 2009 15:01:14 +0100 Subject: [Freeswitch-users] OT: Spa2102 and call transfer In-Reply-To: References: Message-ID: I have the same problem with a HandyTone 502 adapter. Anyone got any hints to get the flash button to work? On Wed, Dec 9, 2009 at 11:25 AM, Jonas Gauffin wrote: > Hello, > > I can't get call transfer to work with a SPA2102 adapter. > I don't think it has something to do with FS, but I'm hoping someone here > could help me. > I do not get a new line in the phone (by pressing the R button), all DTMF > tones are sent as audio to the other connected phone. > > Anyone got it working? > > Thanks, > Jonas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/630a8c1b/attachment.html From mgende at gendesign.com Wed Dec 9 07:16:03 2009 From: mgende at gendesign.com (Michael Gende) Date: Wed, 9 Dec 2009 09:16:03 -0600 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1EB50D020000E1000002C7@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> Message-ID: Hey There, I came in not seeing any former posts of yours, so if this one is unhelpful, just delete. I did my FS install using PFsense as well. Its been working famously for a few months now. Very happy. I wrote what I discovered for FS on PFsense in this wiki: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I assume since you're using PFsense that your computer is functioning as a firewall AND a phone system (a dual homed host, in other words). That's what the wiki attempt above is aimed at. If you follow those instructions, you'll send and receive calls (as we were and are able to). It is working for us, at least. One problem in your case: I didn't really like using the PFsense Web interface when configuring FS (except for installing FS and setting some system parameters. Its great for PFsense, though). It helped me more to get in with ssh and vi and make FS work. Having done that successfully, you'll be more likely to effectively use the PFsense web interface for FS, as it's really just a "short cut" for someone that understands the FS file system, in my opinion. Good luck, Mike G. On Tue, Dec 8, 2009 at 1:20 PM, mailinglist wrote: > Hi All > > Ok, after reading a bit more I think I see what I've done wrong, but I > don't know how to fix it properly. > Looking in the Dialplan directory I see the following: > default (dir) > default.xml > features.xml > public (dir) > public.xml > > Under the default dir the webinterface has created the 001_musimi.dk.xml > file that I've created. > But as I understand it, it doesn't use it. > > How do I make it use it, I would very much like to keep the webinterface > editor, and not have to do it via ssh and vi all the time. > > >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i > meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: > Hi Mark > > Ok, thanks. > Yes I have a gateway placed in external called musimi.dk (or should it be > in public?), and I'll just create the empty XML's in lan to get rid of that > error. > > I'll remove the second part of the dialplan, my idea was that it was needed > for calls between sip phones hooked up to the freeswitch. > > Now the remaining problem: > When I call ext 1002 from ext 1001 I see this message and get an error, the > same goes for dialing 0XXXXXXXX to get an external number: > > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing > 1001->1002 in context default > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 > [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. > Cause: NO_ROUTE_DESTINATION > 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup > sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 > (sofia/external/$1) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/external/$1 [CS_DESTROY] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( > sofia/internal/1001 at 10.11.12.25) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] > I don't see any mention of the statements in the Dialplan, so for me it > looks like it haven't registered the Dialplan? > > Best regards > Kenneth > > >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen > 659603.29094.qm at web56408.mail.re3.yahoo.com: > > Question ---------------------------------------------- > If I do a reloadxml it gives me this output on the console: > freeswitch at firewall.fribert.dk> > reloadxml > 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > > I'm not sure if it's a genuine problem,as I can see it, it just complains > that I haven't created any sip_profiles in /lan, but is that necessary? > > Response: ---------------------------------------------- > This isn't really a problem. To get rid of the error simply put a blank xml > file into each folder as in the internal and external directories. Dump the > lan directory and lan profile as mentioned earlier. > > Question ---------------------------------------------- > > Extension Name musimi.dk > Enabled true > Order 001 > Description ... > > condition ^0(.\d+)$ > action bridge sofia/gateway/musimi.dk/$1 > > Response: ---------------------------------------------- > > This is correct as long as you have a gateway that is registered called > musimi.dk > > Question ---------------------------------------------- > Extension Name 10.11.12.25 > Enabled true > Order 002 > Description ... > > action bridge sofia/internal/$ > > Response: ---------------------------------------------- > > No idea what this is for its not needed as far as I can tell. > > > Now please summarize what you still need help on. > > > Mark J Crane > http://fusionpbx.com > pfSense FreeSWITCH package developer > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/d59b1ce5/attachment.html From nik.middleton at noblesolutions.co.uk Wed Dec 9 07:29:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 9 Dec 2009 15:29:37 -0000 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> Message-ID: I would have tended to agree with the glare, however, before I killed both sides, I was back to my issue of the call not clearing down at all. (rtp timeout eventually does it) Thanks for the pointer to the source. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 09 December 2009 14:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hang-up on B leg src/switch_ivr_bridge.c This could just as well be a glare condition when the call is in process of tearing down. Mike On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote: No doubt, but that's a little difficult as this only happens occasionally and I have 200 calls going on at the time. It's needle in the haystack stuff. Here's what I know. I have an external process listening for DTMF events. If I detect '*' I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn't exist, so I now do a double kill on the associated leg which I get from the event. I do not get a 'doesn't exist' message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. The kluge clears the B leg hang issue, so the pressure's off for me, but when I get a few nano seconds, I'll look at the code to see if there's anything obvious. Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 08 December 2009 16:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/e391ec1a/attachment-0001.html From fernando.testa at gmail.com Wed Dec 9 07:25:11 2009 From: fernando.testa at gmail.com (Fernando Testa) Date: Wed, 9 Dec 2009 13:25:11 -0200 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones In-Reply-To: References: Message-ID: <22F7B75B-6568-4863-A7ED-165A800A928D@gmail.com> It worked! Tnx! Em 08/12/2009, ?s 16:51, Brian West escreveu: > Best option for you is to use 96 in the sofia profile you're using to > talk to these broken devices. > > /b > > On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: > >> Dear list, >> >> Some Nec phones sends DTMF RFC2833 with payload 101 during the call, >> but have negotiated a different one on SDP. >> When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 >> we notice this phone sends the following INVITE packet and RTP >> packets: http://pastebin.freeswitch.org/11433 >> Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap >> >> Is there any parameter to tweak FS in such a way to force understand >> 101 packets as DTMF? >> Thank you in advance! >> >> Fernando Testa >> PS: On pcap you have the following IPs: >> FS at 10.91.10.210 >> Nec Pbx 10.91.10.22 >> phone 10.91.10.85 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lgraybill at izeni.com Wed Dec 9 08:50:03 2009 From: lgraybill at izeni.com (Luke Graybill) Date: Wed, 9 Dec 2009 09:50:03 -0700 Subject: [Freeswitch-users] controlling calls handled within a fifo using event_socket Message-ID: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> In my FreeSWITCH environment, calls are originated out to customers who are placed into a fifo upon answer. There are members (x-lite endpoints) in this fifo who handle those customer calls. I am writing a monitoring application that uses event_socket to watch the channels involved in this process, ultimately displaying an interface for each rep that allows them to interactively drive the calls (playback audio conditionally to the customer, save information obtained during the call to another database, etc). Problems arise when attempting to identify which customer channel is speaking to which rep (consumer) channel. My event_socket application is inspecting the CHANNEL_ANSWER event, but this event does not appear to contain enough information to make this determination. I have identified three distinct uuid values in the CHANNEL_ANSWER headers on the consumer channel: core uuid, the uuid of the consumer channel, and another uuid which is not the customer uuid (I'm assuming this is the uuid of the fifo). According to the wiki here, I expected the consumer CHANNEL_ANSWER headers to contain variables such as `fifo_target` with the uuid of the customer channel it is bridged to, but this variable is not in the headers. Indeed, no channel variables are set which correspond to the uuid of the customer channel to which the rep is speaking. After the call has been completed, data posted in the cdr does in fact contain the `fifo_target` information, but this does not help me during the call. The short version of my question is this: how do I programmatically determine which channel uuid the consumer channel in a fifo is connected to? Any help here would be greatly appreciated :) Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/5aed75f9/attachment.html From brian at freeswitch.org Wed Dec 9 08:55:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 10:55:36 -0600 Subject: [Freeswitch-users] controlling calls handled within a fifo using event_socket In-Reply-To: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> References: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> Message-ID: <68510DE6-DBCF-406E-92E9-8C67B29AF59F@freeswitch.org> "fifo list" issue this API and get the fifo XML and get the caller's uuid out of the list. /b On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote: > The short version of my question is this: how do I programmatically > determine which channel uuid the consumer channel in a fifo is > connected to? From Prometheus001 at gmx.net Wed Dec 9 09:13:55 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 09 Dec 2009 18:13:55 +0100 Subject: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer Message-ID: <4B1FDAD3.8020305@gmx.net> Hello, in our dialplan we have enabled multiple-registrations, so 2 phones can register on a single directory entry. Both phones are registered, both phones can be called and each phone can call the other phone. However in an attended_transfer mode calls cannot be transferred to the other phone with the same number. Attended_transfer in this case is needed when you take a call on your main SIP phone and and then want to transfer it to your mobile DECT/SIP phone, because you may have to check something in another room. I did a SIP trace and see the following: * A invites B(phone 1) => ok * B(phone 1) places call on hold => ok * B(phone 1) dials number B(phone 2 DECT) on second line * Freeswitch send Invite to B(phone 1) => ok * Freeswitch send Invite to B(phone 2 DECT) * B(phone 2 DECT) sends Ringing to Freeswitch => ok * B(phone 1) sends Busy to Freeswitch * B(phone 1) displays Busy and hangs up the second line Is there any way to overcome this? Is there a way to ignore the Busy from phone 1 when phone 2 answers Ringing? Best regards Peter From edpimentl at gmail.com Wed Dec 9 09:29:11 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 9 Dec 2009 12:29:11 -0500 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> References: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> Message-ID: <9dc4a1670912090929x1193c69du6cb47ec4da479b39@mail.gmail.com> Regarding Mac OSX 10.5/6 can you point me where the latest "FS binary" file is? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/62930daf/attachment.html From msc at freeswitch.org Wed Dec 9 09:41:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Dec 2009 09:41:21 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: Message-ID: <87f2f3b90912090941p245d788cr4c530e17f88e162f@mail.gmail.com> On Tue, Dec 8, 2009 at 3:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Thought I?d send this little hurrah! As there seems to have been a lot > of negativity on this list lately. > > > I hereby multiply all the negative comments by -1. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/36dccb24/attachment.html From jmesquita at freeswitch.org Wed Dec 9 10:37:37 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 9 Dec 2009 16:37:37 -0200 Subject: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer In-Reply-To: <4B1FDAD3.8020305@gmx.net> References: <4B1FDAD3.8020305@gmx.net> Message-ID: That is more dependent on the endpoint than on the switch itself. I guess you can always use mod_limit to come up with some crazy key to identify one endpoint or the other but still it seems overly complicated for something that is not supposed to be working this way. You can also park the call instead of transferring, can't ya? JM On Wed, Dec 9, 2009 at 3:13 PM, Peter P GMX wrote: > Hello, > > in our dialplan we have enabled multiple-registrations, so 2 phones can > register on a single directory entry. > > Both phones are registered, both phones can be called and each phone can > call the other phone. > However in an attended_transfer mode calls cannot be transferred to the > other phone with the same number. > Attended_transfer in this case is needed when you take a call on your > main SIP phone and and then want to transfer it to your mobile DECT/SIP > phone, because you may have to check something in another room. > I did a SIP trace and see the following: > > * A invites B(phone 1) => ok > * B(phone 1) places call on hold => ok > * B(phone 1) dials number B(phone 2 DECT) on second line > * Freeswitch send Invite to B(phone 1) => ok > * Freeswitch send Invite to B(phone 2 DECT) > * B(phone 2 DECT) sends Ringing to Freeswitch => ok > * B(phone 1) sends Busy to Freeswitch > * B(phone 1) displays Busy and hangs up the second line > > Is there any way to overcome this? Is there a way to ignore the Busy > from phone 1 when phone 2 answers Ringing? > > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/3c76a285/attachment.html From codecomplete at free.fr Wed Dec 9 11:54:51 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 9 Dec 2009 11:54:51 -0800 (PST) Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: Message-ID: <26716612.post@talk.nabble.com> Nik Middleton wrote: > I cannot imagine doing what I'm using FS for, with any other product. Yes > it's frustrating at times, but this is largely down to a lack > documentation/samples. Speaking of which... would this layout be good for a book on Freeswitch? Preface 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. 2. Choosing hardware options (server, phones, gateways) 3. Setting up FS 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS gateways, etc.) 5. Administering FS (CLI and GUI) 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) 7. Performance, sound quality, other issues 8. Writing scripts (LUA, etc.), connecting to databases 9. Real-life examples (Gino's Pizza, etc.) Conclusion Index -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p26716612.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dftoro at yahoo.com Wed Dec 9 12:04:56 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 9 Dec 2009 12:04:56 -0800 (PST) Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: Message-ID: <771079.9902.qm@web33508.mail.mud.yahoo.com> I had hear about Welltech (http://www.welltech.com/default.aspx) gateways but I don't have any experience with them. Someone know ?, any experience... Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 11/25/09, Milena wrote: From: Milena Subject: Re: [Freeswitch-users] Grandstream gateways To: freeswitch-users at lists.freeswitch.org Date: Wednesday, November 25, 2009, 4:00 PM Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. ?I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. ?If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. ?If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. ?Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. ?Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? ?Please share your experience > with my boss, ?Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. ?I wanted assistance with choosing the most appropriate > hardware. ?We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 ?FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > ?wrote: > >> Send FreeSWITCH-users mailing list submissions to >> ? freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> ? freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> ? freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> ?1. Re: mod_conference kick to abort invitations (Michael Jerris) >> ?2. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Michael Jerris) >> ?3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> ?4. Re: remote_media_ip variable not set (Michael Jerris) >> ?5. Re: How to find whether the destination ? ?extension supports >> ? ? encryption (Michael Jerris) >> ?6. Re: Bypass_media and re_invite (srinivasula reddy) >> ?7. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Stephen Crosby) >> ?8. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> ? invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an ?invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. ?This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> ? proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. ?I just added some code to set it on auto- >> adjust so it might be there sometimes now. ?You might need to add >> some code in mod_sofia to add it other times. ?Maybe it makes sense >> to move that var setting down to switch_rtp.c. ?Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> ? extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the ?first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. ?i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. ?This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> ? >> ? ? > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> ? ? ? ? >> ? ? ? ? >> >> ? ? ? ?> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> ? ? ? ?> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> ? ? ? ?> data="aPfx=${caller_id_number:0:6}"/> >> ? ? ? ?> data="aNum=${caller_id_number:6:16}"/> >> ? ? ? ?> data="IP_ADDR=${network_addr}:5060"/> >> >> ? ? ? ? >> >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> >> ? ? ? ? >> ? ? ? >> ? >> >> >> ? >> ? ? >> ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? >> >> ? >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. ?This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/1bf7a2f7/attachment-0001.html From ken at ksac.com Wed Dec 9 12:12:36 2009 From: ken at ksac.com (Kendall Stauffer) Date: Wed, 9 Dec 2009 12:12:36 -0800 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <771079.9902.qm@web33508.mail.mud.yahoo.com> References: <771079.9902.qm@web33508.mail.mud.yahoo.com> Message-ID: Yes. I have one if anybody wants it, would let it go cheap. Works fine, but caller id is only the number, not the name part. Other than that works fine with astersik From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Toro Sent: Wednesday, December 09, 2009 3:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Grandstream gateways I had hear about Welltech (http://www.welltech.com/default.aspx) gateways but I don't have any experience with them. Someone know ?, any experience... Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 11/25/09, Milena wrote: From: Milena Subject: Re: [Freeswitch-users] Grandstream gateways To: freeswitch-users at lists.freeswitch.org Date: Wednesday, November 25, 2009, 4:00 PM Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen > You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti > wrote: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews > wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> > >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/751d0ab1/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Dec 9 12:50:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 9 Dec 2009 20:50:01 -0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26716612.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> Message-ID: Looks good, but you've missed out billing and the key one, the event socket which could be a chapter in it's self. Do you have a publisher for it yet? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 Sent: 09 December 2009 19:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Rocks!!!!!!!!! Nik Middleton wrote: > I cannot imagine doing what I'm using FS for, with any other product. Yes > it's frustrating at times, but this is largely down to a lack > documentation/samples. Speaking of which... would this layout be good for a book on Freeswitch? Preface 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. 2. Choosing hardware options (server, phones, gateways) 3. Setting up FS 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS gateways, etc.) 5. Administering FS (CLI and GUI) 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) 7. Performance, sound quality, other issues 8. Writing scripts (LUA, etc.), connecting to databases 9. Real-life examples (Gino's Pizza, etc.) Conclusion Index -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p267 16612.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mailinglist at fribert.dk Wed Dec 9 13:46:00 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:46:00 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through Message-ID: <4B2028A8020000E10000030E@mail.fribert.dk> Hi Michael Thankyou for the excellent wiki article, yes, I did follow your guide there, all the way except to 'dialplan' and it seems that's the problem at the moment. I would very much like to create the dialplan in the webinterface, and not in the public.xml file. But at the moment it only uses the public.xml file :-( Great writeup you made, and it has brought me a long way. BR Fribert >>> 09-12-2009 kl. 16:16 skrev Michael Gende i meddelelsen : Hey There, I came in not seeing any former posts of yours, so if this one is unhelpful, just delete. I did my FS install using PFsense as well. Its been working famously for a few months now. Very happy. I wrote what I discovered for FS on PFsense in this wiki: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I assume since you're using PFsense that your computer is functioning as a firewall AND a phone system (a dual homed host, in other words). That's what the wiki attempt above is aimed at. If you follow those instructions, you'll send and receive calls (as we were and are able to). It is working for us, at least. One problem in your case: I didn't really like using the PFsense Web interface when configuring FS (except for installing FS and setting some system parameters. Its great for PFsense, though). It helped me more to get in with ssh and vi and make FS work. Having done that successfully, you'll be more likely to effectively use the PFsense web interface for FS, as it's really just a "short cut" for someone that understands the FS file system, in my opinion. Good luck, Mike G. On Tue, Dec 8, 2009 at 1:20 PM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/32371c67/attachment.html From mailinglist at fribert.dk Wed Dec 9 13:47:04 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:47:04 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <187489.95329.qm@web56408.mail.re3.yahoo.com> References: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> <187489.95329.qm@web56408.mail.re3.yahoo.com> Message-ID: <4B2028E8020000E100000313@mail.fribert.dk> This is a new install, but it's grabbed from a pfSense repository. >>> 09-12-2009 kl. 10:28 skrev Mark Crane i meddelelsen <187489.95329.qm at web56408.mail.re3.yahoo.com>: Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch at yahoo.com --- On Tue, 12/8/09, Nandy Dagondon wrote: From: Nandy Dagondon Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk ( /mc/compose?to=4B1DFABC020000E1000002C2 at mail.fribert.dk )>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 )) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com ( /mc/compose?to=659603.29094.qm at web56408.mail.re3.yahoo.com ): Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/bedc27ab/attachment-0001.html From mailinglist at fribert.dk Wed Dec 9 13:47:52 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:47:52 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <4B202918020000E100000318@mail.fribert.dk> Yes, I have two extensions. I can even make them join a group, and if I call the group, the two extensions will ring. >>> 08-12-2009 kl. 23:45 skrev Nandy Dagondon i meddelelsen <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4 at mail.gmail.com>: have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/58f3afc3/attachment.html From timuckun at gmail.com Wed Dec 9 13:56:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 10:56:31 +1300 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26716612.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> Message-ID: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> > > Preface > 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. > 2. Choosing hardware options (server, phones, gateways) > 3. Setting up FS > 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS > gateways, etc.) > 5. Administering FS (CLI and GUI) > 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) > 7. Performance, sound quality, other issues > 8. Writing scripts (LUA, etc.), connecting to databases > 9. Real-life examples (Gino's Pizza, etc.) > Conclusion > Index > -- I found the rosetta stone useful though woefully lacking in volume. I guess that's true overall with the project. From brian at freeswitch.org Wed Dec 9 14:07:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 16:07:28 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> Message-ID: <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> Visit the friday meetings and we can help if you document it. ;) /b On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote: > I found the rosetta stone useful though woefully lacking in volume. > > I guess that's true overall with the project. From msc at freeswitch.org Wed Dec 9 14:10:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Dec 2009 14:10:35 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> Message-ID: <87f2f3b90912091410p322a267cne481079856f74ed5@mail.gmail.com> > I found the rosetta stone useful though woefully lacking in volume. > > I guess that's true overall with the project. > > Documentation is neither easy nor glamorous. The woefully lacking documentation has been provided by a little group of people who've done a big bit of documenting and a big group of people who've done a little bit of documenting. If ever there was an aspect of this project that could use more volunteers it is documentation and bug testing. If anyone wants to help on either of these fronts please email me off list. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/5906c1be/attachment.html From mailinglist at fribert.dk Wed Dec 9 14:20:01 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 23:20:01 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <4B2030A1020000E10000031D@mail.fribert.dk> WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/6f3edd80/attachment-0001.html From timuckun at gmail.com Wed Dec 9 14:35:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 11:35:31 +1300 Subject: [Freeswitch-users] Even socket question. Message-ID: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> Hey All. I am trying to get freeswitch to route to my socket handler and am having a problem. I am running freeswitch inside a virtualbox VM for testing purposes. The vitualbox communicates with my host via the "host only" adapter. The VM IP address is 192.168.56.3 and the laptop has the iP 192.168.56.1 I have set up both an outbound and an inbound socket handlers. The inbound one works fine, the outbound is not working . The inbound merely logs the event name. The outbound logs the connection and hangs up. I have set up an extension like this When I dial 8084 I get a lot of events being logged but the oubound never gets the calls and never logs the call. I have added the fs_cli output below. It looks to me like it's sending the output to the other IP address of my laptop instead of the one I specified in my extension but I could just be misreading that. I have set the external IP of the freeswitch to the 56.3 address. Here is the LSOF output freeswitc 2468 root 31u IPv4 5785 TCP ubuntuvm01:5080 (LISTEN) freeswitc 2468 root 33u IPv6 5791 TCP localhost:5060 (LISTEN) freeswitc 2468 root 36u IPv4 5804 TCP 192.168.56.3:5060 (LISTEN) freeswitc 2468 root 48u IPv4 5910 TCP 192.168.56.3:8021 (LISTEN) freeswitc 2468 root 50u IPv4 5912 TCP *:8080 (LISTEN) Here is the output from the fs_cli 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl "domains". Falling back to Digest auth. 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl "domains". Falling back to Digest auth. 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1000 at 192.168.56.3 [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa] 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_NEW 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.56.3) State NEW 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [received][100] 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP: v=0 o=Z 0 0 IN IP4 218.101.6.157 s=Z c=IN IP4 218.101.6.157 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:115:32000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:107:16000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G722:9:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMU:0:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMA:8:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec sofia/internal/1000 at 192.168.56.3 GSM/8000 20 ms 160 samples 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload to 101 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885 (sofia/internal/1000 at 192.168.56.3) State Change CS_NEW -> CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.56.3) State INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.56.3 SOFIA INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111 (sofia/internal/1000 at 192.168.56.3) State Change CS_INIT -> CS_ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.56.3) State INIT going to sleep 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.56.3) State ROUTING 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:132 sofia/internal/1000 at 192.168.56.3 SOFIA ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1000 at 192.168.56.3 Standard ROUTING 2009-12-09 14:31:53.423898 [INFO] mod_dialplan_xml.c:408 Processing 1000->8084 in context default Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->tod_example] continue=true Dialplan: day of week[4] =~ 2-6 (PASS) Dialplan: hour[14] =~ 9-18 (PASS) Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Action set(open=true) Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->holiday_example] continue=true Dialplan: month[12] =~ 1 (FAIL) Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global-intercept] destination_number(8084) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [group-intercept] destination_number(8084) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [intercept-ext] destination_number(8084) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [redial] destination_number(8084) =~ /^870$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 192.168.56.3 Absolute Condition [global] Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-2] destination_number(8084) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-1] destination_number(8084) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] destination_number(8084) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] destination_number(8084) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [call_return] destination_number(8084) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [del-group] destination_number(8084) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Action answer() Dialplan: sofia/internal/1000 at 192.168.56.3 Action group(delete:84@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action gentones(%(1000, 0, 320)) 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:122 (sofia/internal/1000 at 192.168.56.3) State Change CS_ROUTING -> CS_EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.56.3) State ROUTING going to sleep 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.56.3) State EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:181 sofia/internal/1000 at 192.168.56.3 SOFIA EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:159 sofia/internal/1000 at 192.168.56.3 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.56.3 set(open=true) 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:768 sofia/internal/1000 at 192.168.56.3 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-spymap/1000/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-last_dial/1000/8084) EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-last_dial/global/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) EXECUTE sofia/internal/1000 at 192.168.56.3 answer() 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:658 sofia/internal/1000 at 192.168.56.3 receive message [ANSWER] 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:2381 AUDIO RTP [sofia/internal/1000 at 192.168.56.3] 192.168.50.173 port 27042 -> 218.101.6.157 port 8000 codec: 3 ms: 20 2009-12-09 14:31:53.423898 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 160 bytes per 20ms 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:571 Local SDP sofia/internal/1000 at 192.168.56.3: v=0 o=FreeSWITCH 1260370871 1260370872 IN IP4 192.168.50.173 s=FreeSWITCH c=IN IP4 192.168.50.173 t=0 0 m=audio 27042 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [NOTICE] mod_dptools.c:658 Channel [sofia/internal/1000 at 192.168.56.3] has been answered 2009-12-09 14:31:53.423898 [DEBUG] switch_channel.c:182 sofia/internal/1000 at 192.168.56.3 receive message [AUDIO_SYNC] EXECUTE sofia/internal/1000 at 192.168.56.3 group(delete:84 at 192.168.56.3:sofia/internal/sip:1000 at 218.101.6.157:5070;rinstance=a8b6fdbc731e3b66;transport=UDP) EXECUTE sofia/internal/1000 at 192.168.56.3 gentones(%(1000, 0, 320)) 2009-12-09 14:31:53.436374 [DEBUG] switch_core_io.c:652 sofia/internal/1000 at 192.168.56.3 receive message [TRANSCODING_NECESSARY] 2009-12-09 14:31:53.436670 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [completed][200] 2009-12-09 14:31:53.490803 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [ready][200] 2009-12-09 14:31:53.729534 [INFO] switch_rtp.c:1987 Auto Changing port from 218.101.6.157:8000 to 192.168.50.105:8000 2009-12-09 14:31:54.430526 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1000 at 192.168.56.3 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-09 14:31:54.430526 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/1000 at 192.168.56.3 [KILL] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 (sofia/internal/1000 at 192.168.56.3) State HANGUP 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:358 Channel sofia/internal/1000 at 192.168.56.3 hanging up, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:400 Sending BYE to sofia/internal/1000 at 192.168.56.3 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 at 192.168.56.3 Standard HANGUP, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 (sofia/internal/1000 at 192.168.56.3) State HANGUP going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.56.3) State EXECUTE going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_HANGUP 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:465 sofia/internal/1000 at 192.168.56.3 handler already called, skipping state handler. 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000 at 192.168.56.3) State Change CS_HANGUP -> CS_REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/1000 at 192.168.56.3) State REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 at 192.168.56.3 Standard REPORTING, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/1000 at 192.168.56.3) State REPORTING going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1000 at 192.168.56.3) State Change CS_REPORTING -> CS_DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:1136 Session 6 (sofia/internal/1000 at 192.168.56.3) Locked, Waiting on external entities 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/internal/1000 at 192.168.56.3) Ended 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1156 Close Channel sofia/internal/1000 at 192.168.56.3 [CS_DESTROY] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 192.168.56.3) State DESTROY 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:293 sofia/internal/1000 at 192.168.56.3 SOFIA DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 192.168.56.3 Standard DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 192.168.56.3) State DESTROY going to sleep From timuckun at gmail.com Wed Dec 9 14:39:26 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 11:39:26 +1300 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> Message-ID: <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> On Thu, Dec 10, 2009 at 11:07 AM, Brian West wrote: > Visit the friday meetings and we can help if you document it. ?;) > I would be willing to lend a hand with the documentation but I know so little (a complete freeswitch noob). For example I was trying to figure out how to tell if an extension was set up "show dialplan in asterisk". I could not find this anywhere. If I find out I would be happy to add it to the rosetta stone. I am currently working on getting outbound socket working. Once I get it going I would be happy to add it to the relevant section of the wiki (in this case ruby). From brian at freeswitch.org Wed Dec 9 14:43:34 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 16:43:34 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> Message-ID: <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> That is what is nice about our community I'm more than willing to answer the questions if you document them... as are many others in the core team...we just have a lot to do and I think the best repayment is documentation! ;) /b On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > On Thu, Dec 10, 2009 at 11:07 AM, Brian West > wrote: >> Visit the friday meetings and we can help if you document it. ;) >> > > I would be willing to lend a hand with the documentation but I know so > little (a complete freeswitch noob). For example I was trying to > figure out how to tell if an extension was set up "show dialplan in > asterisk". I could not find this anywhere. If I find out I would be > happy to add it to the rosetta stone. > > I am currently working on getting outbound socket working. Once I get > it going I would be happy to add it to the relevant section of the > wiki (in this case ruby). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/f0155fb5/attachment.html From mctch at yahoo.com Wed Dec 9 15:00:36 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 15:00:36 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B2030A1020000E10000031D@mail.fribert.dk> Message-ID: <110796.60596.qm@web56401.mail.re3.yahoo.com> Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. ????? Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! ? It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? ? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml ? ? The default.xml looks like this ( I haven't changed it): ? ? ? ??? ????? ????? ? ????? ??? ? ??? ??? ????? ????? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ????? ? ????? ????? ????? ????? ? ? ? ????? ? ????? ? ? ? ????? ??? ? ??? ??? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ??? ??? ?? ??? ??? ????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ? ? ??? ??? ????? ? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ??? ????? ? ????? ??? ? ??? ?????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ??????? ??????? ? ????? ??? ? ??? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ????? ? ????? ??? ??? ??? ????? ????? ????? ? ? ????? ??? ? ??? ??? ????? ????? ????? ?]]> ? ????? ??? ??? ??? ??? ????? ????? ????? ? ? ?????? ??? ? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ??????? ??????? ????? ??? ? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ? ??? ????? ????? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ??? ??? ? ? ??? ??? ????? ??? ? ? ??? ??? ??? ? ??? ? ? ? ? Then I have under default dir: musimidk.xml ?? ?????? ?? and 9000_recordings.xml ?? ?????? ?? ? ? Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/6e5a4446/attachment-0001.html From brian at freeswitch.org Wed Dec 9 15:03:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 17:03:02 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Bug Reports... Message-ID: Dear FreeSWITCHers, As of Friday Dec. 11th we will NOT accept any more bug reports on 1.0.4. You need to be on a 1.0.5pre or SVN trunk. 1.0.4 is over 6 months old and I really suspect your issues in 1.0.4 are already fixed. We will release a new pre every monday morning till 1.0.5 is released please keep up to date if possible. We are working hard to get 1.0.5 out and be as stable as possible and its more stable than 1.0.4... their might be some edge or corner cases that aren't accounted for so we need you to please download SVN trunk in your test labs and try it out... report issues and help us make the best FreeSWITCH release possible. Thank you, Brian West From djbinter at yahoo.com Wed Dec 9 15:07:32 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 9 Dec 2009 15:07:32 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> Message-ID: <769416.34727.qm@web37508.mail.mud.yahoo.com> Load sharing feature is coming off our Lucent Telica switch. ________________________________ From: Tim Uckun To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 9, 2009 2:26:41 AM Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. On Tue, Dec 8, 2009 at 5:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > Any suggestion would be appreciated. Thank you. If you could explain how you are doing the load balancing it would be really helpful to me. I am trying to do the same thing. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/75713a81/attachment.html From anthony.minessale at gmail.com Wed Dec 9 17:46:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Dec 2009 19:46:07 -0600 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> Message-ID: <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> do you have something listening on 8084 ? On Wed, Dec 9, 2009 at 4:35 PM, Tim Uckun wrote: > Hey All. I am trying to get freeswitch to route to my socket handler > and am having a problem. > > I am running freeswitch inside a virtualbox VM for testing purposes. > The vitualbox communicates with my host via the "host only" adapter. > The VM IP address is 192.168.56.3 and the laptop has the iP > 192.168.56.1 > > I have set up both an outbound and an inbound socket handlers. The > inbound one works fine, the outbound is not working . The inbound > merely logs the event name. The outbound logs the connection and hangs > up. > > I have set up an extension like this > > > > > > > > > > > > > When I dial 8084 I get a lot of events being logged but the oubound > never gets the calls and never logs the call. > > I have added the fs_cli output below. It looks to me like it's sending > the output to the other IP address of my laptop instead of the one I > specified in my extension but I could just be misreading that. I > have set the external IP of the freeswitch to the 56.3 address. > > Here is the LSOF output > > freeswitc 2468 root 31u IPv4 5785 > TCP ubuntuvm01:5080 (LISTEN) > freeswitc 2468 root 33u IPv6 5791 > TCP localhost:5060 (LISTEN) > freeswitc 2468 root 36u IPv4 5804 > TCP 192.168.56.3:5060 (LISTEN) > freeswitc 2468 root 48u IPv4 5910 > TCP 192.168.56.3:8021 (LISTEN) > freeswitc 2468 root 50u IPv4 5912 > TCP *:8080 (LISTEN) > > > Here is the output from the fs_cli > > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0] > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0] > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/1000 at 192.168.56.3 > [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa] > 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_NEW > 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/1000 at 192.168.56.3) State NEW > 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [received][100] > 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP: > v=0 > o=Z 0 0 IN IP4 218.101.6.157 > s=Z > c=IN IP4 218.101.6.157 > t=0 0 > m=audio 8000 RTP/AVP 3 110 98 8 0 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=30 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G7221:115:32000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G7221:107:16000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G722:9:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[PCMU:0:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[PCMA:8:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[GSM:3:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec > sofia/internal/1000 at 192.168.56.3 GSM/8000 20 ms 160 samples > 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf > payload to 101 > 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885 > (sofia/internal/1000 at 192.168.56.3) State Change CS_NEW -> CS_INIT > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_INIT > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.56.3) State INIT > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83 > sofia/internal/1000 at 192.168.56.3 SOFIA INIT > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111 > (sofia/internal/1000 at 192.168.56.3) State Change CS_INIT -> CS_ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.56.3) State INIT going to sleep > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.56.3) State ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:132 > sofia/internal/1000 at 192.168.56.3 SOFIA ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1000 at 192.168.56.3 Standard ROUTING > 2009-12-09 14:31:53.423898 [INFO] mod_dialplan_xml.c:408 Processing > 1000->8084 in context default > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Action set(open=true) > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [global-intercept] destination_number(8084) =~ /^886$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [group-intercept] destination_number(8084) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [intercept-ext] destination_number(8084) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->redial] > continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [redial] > destination_number(8084) =~ /^870$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global] > continue=true > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: sofia/internal/1000 at 192.168.56.3 Absolute Condition [global] > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->snom-demo-2] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-2] > destination_number(8084) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->snom-demo-1] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-1] > destination_number(8084) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] > destination_number(8084) =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] > destination_number(8084) =~ /^779$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->call_return] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [call_return] > destination_number(8084) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->del-group] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [del-group] > destination_number(8084) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Action answer() > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > group(delete:84@${domain_name}:${sofia_contact(${sip_from_user}@ > ${domain_name})}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action gentones(%(1000, 0, > 320)) > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/1000 at 192.168.56.3) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.56.3) State ROUTING going to sleep > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.56.3) State EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:181 > sofia/internal/1000 at 192.168.56.3 SOFIA EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/1000 at 192.168.56.3 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.56.3 set(open=true) > 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:768 > sofia/internal/1000 at 192.168.56.3 SET [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.56.3 > hash(insert/192.168.56.3-spymap/1000/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) > EXECUTE sofia/internal/1000 at 192.168.56.3 > hash(insert/192.168.56.3-last_dial/1000/8084) > EXECUTE sofia/internal/1000 at 192.168.56.3 > > hash(insert/192.168.56.3-last_dial/global/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) > EXECUTE sofia/internal/1000 at 192.168.56.3 answer() > 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:658 > sofia/internal/1000 at 192.168.56.3 receive message [ANSWER] > 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:2381 AUDIO RTP > [sofia/internal/1000 at 192.168.56.3] 192.168.50.173 port 27042 -> > 218.101.6.157 port 8000 codec: 3 ms: 20 > 2009-12-09 14:31:53.423898 [DEBUG] switch_rtp.c:1167 Starting timer > [soft] 160 bytes per 20ms > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:571 Local SDP > sofia/internal/1000 at 192.168.56.3: > v=0 > o=FreeSWITCH 1260370871 1260370872 IN IP4 192.168.50.173 > s=FreeSWITCH > c=IN IP4 192.168.50.173 > t=0 0 > m=audio 27042 RTP/AVP 3 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:645 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [NOTICE] mod_dptools.c:658 Channel > [sofia/internal/1000 at 192.168.56.3] has been answered > 2009-12-09 14:31:53.423898 [DEBUG] switch_channel.c:182 > sofia/internal/1000 at 192.168.56.3 receive message [AUDIO_SYNC] > EXECUTE sofia/internal/1000 at 192.168.56.3 > group(delete:84 at 192.168.56.3:sofia/internal/sip:1000 at 218.101.6.157:5070 > ;rinstance=a8b6fdbc731e3b66;transport=UDP) > EXECUTE sofia/internal/1000 at 192.168.56.3 gentones(%(1000, 0, 320)) > 2009-12-09 14:31:53.436374 [DEBUG] switch_core_io.c:652 > sofia/internal/1000 at 192.168.56.3 receive message > [TRANSCODING_NECESSARY] > 2009-12-09 14:31:53.436670 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [completed][200] > 2009-12-09 14:31:53.490803 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [ready][200] > 2009-12-09 14:31:53.729534 [INFO] switch_rtp.c:1987 Auto Changing port > from 218.101.6.157:8000 to 192.168.50.105:8000 > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_state_machine.c:187 > Hangup sofia/internal/1000 at 192.168.56.3 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-09 14:31:54.430526 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/1000 at 192.168.56.3 [KILL] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 > (sofia/internal/1000 at 192.168.56.3) State HANGUP > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/1000 at 192.168.56.3 hanging up, cause: NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:400 Sending BYE to > sofia/internal/1000 at 192.168.56.3 > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1000 at 192.168.56.3 Standard HANGUP, cause: > NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 > (sofia/internal/1000 at 192.168.56.3) State HANGUP going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.56.3) State EXECUTE going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_HANGUP > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:465 > sofia/internal/1000 at 192.168.56.3 handler already called, skipping > state handler. > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/1000 at 192.168.56.3) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 > (sofia/internal/1000 at 192.168.56.3) State REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/1000 at 192.168.56.3 Standard REPORTING, cause: > NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 > (sofia/internal/1000 at 192.168.56.3) State REPORTING going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/1000 at 192.168.56.3) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:1136 Session > 6 (sofia/internal/1000 at 192.168.56.3) Locked, Waiting on external > entities > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1154 Session > 6 (sofia/internal/1000 at 192.168.56.3) Ended > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1156 Close > Channel sofia/internal/1000 at 192.168.56.3 [CS_DESTROY] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/1000 at 192.168.56.3) State DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:293 > sofia/internal/1000 at 192.168.56.3 SOFIA DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/1000 at 192.168.56.3 Standard DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/1000 at 192.168.56.3) State DESTROY going to sleep > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/1c3e9f15/attachment-0001.html From timuckun at gmail.com Wed Dec 9 18:00:23 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 15:00:23 +1300 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> Message-ID: <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale wrote: > do you have something listening on 8084 ? > Yes. I figured out the problem. There was already an extension called 8084 and it overwrote the extension I defined. Which brings me back to a question I had earlier. Where is the equivalent of the "show dialplan" command? How can I list all the extensions and their definitions? From anthony.minessale at gmail.com Wed Dec 9 18:12:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Dec 2009 20:12:07 -0600 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> Message-ID: <191c3a030912091812o5122b5cayd7e82a37ac224a43@mail.gmail.com> the dialplan is dynamic there is no such thing you have to look in your dialplan xml files because it's served up live. FS has a different paradigm than asterisk. On Wed, Dec 9, 2009 at 8:00 PM, Tim Uckun wrote: > On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale > wrote: > > do you have something listening on 8084 ? > > > > Yes. > > I figured out the problem. There was already an extension called 8084 > and it overwrote the extension I defined. > > Which brings me back to a question I had earlier. > > Where is the equivalent of the "show dialplan" command? How can I list > all the extensions and their definitions? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/99e3aa86/attachment.html From brian at microcomaustralia.com.au Wed Dec 9 16:55:45 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 11:55:45 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware Message-ID: <20091210005545.GD28041@sys11.in.vpac.org> Hello, I asked this question on my local linux user group mailing list, and got the recommendation to ask here. Anyway, at the moment I am running Asterisk on an IP04 embedded system. http://www.rowetel.com/ucasterisk/ip04.html It works well most of the time, however there are some bugs that do, under circumstances lead to less then desirable behaviour (such as on some occasions which I don't fully understand sometimes the remote system fails to generate any audio packets when there is no audio - almost like silence suppression was supported by the remote system - and asterisk fails to generate any audio packets in return; on another slower computer running the same SIP software and on the same network everything works fine; as far as I can tell the software - twinkle - doesn't even support silence suppression). I suspect at least some - if not all - of the issues I have encountered may be resolved with Freeswitch, however I don't really want to replace my small, energy efficient, embedded system, with a large, power hungry computer system. Overkill. An added complication is I need at least 1 analogue port to connect to the Australian based telephone line (2 ports exchange ports and 1 extension port would be ideal but not essiential). Unfortunately, I have been told that the IP04 hardware isn't compatable with the requirements of Freeswitch. Such as not having a MMU. So there doesn't appear to be much effort porting Freeswitch to IP04 as a result. I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Are there any other good alternatives? Thanks. -- Brian May From kristian.kielhofner at gmail.com Wed Dec 9 19:47:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 9 Dec 2009 22:47:05 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Brian, I have been making efforts to fully support FreeSWITCH in AstLinux. Our primary targets are low powered x86 boards like the Soekris and Alix. x86, powerful enough, cheap enough (as low as $100), and about 12 watts. Not bad. The Soekris net5501 and standard case will (I believe) take a full height card. Then again you could use any board and get an external SIP gateway (ATA). We don't currently support OpenZAP with FS in AstLinux but I'd love to add support for it eventually. I'm currently working with the FS devs on getting some issues in trunk resolved to get cross compiling working again. Until then you can find ISOs with FreeSWITCH and AstLInux here if you'd like to check it out: http://mirror.astlinux.org/freeswitch/daily/ Let me know what you think. On Wed, Dec 9, 2009 at 7:55 PM, Brian May wrote: > Hello, > > I asked this question on my local linux user group mailing list, and got the > recommendation to ask here. > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > http://www.rowetel.com/ucasterisk/ip04.html > > It works well most of the time, however there are some bugs that do, under > circumstances lead to less then desirable behaviour (such as on some occasions > which I don't fully understand sometimes the remote system fails to generate > any audio packets when there is no audio - almost like silence suppression was > supported by the remote system - and asterisk fails to generate any audio > packets in return; on another slower computer running the same SIP software and > on the same network everything works fine; as far as I can tell the software - > twinkle - doesn't even support silence suppression). > > I suspect at least some - if not all - of the issues I have encountered may be > resolved with Freeswitch, however I don't really want to replace my small, > energy efficient, embedded system, with a large, power hungry computer system. > Overkill. > > An added complication is I need at least 1 analogue port to connect to the > Australian based telephone line (2 ports exchange ports and 1 extension port > would be ideal but not essiential). > > Unfortunately, I have been told that the IP04 hardware isn't compatable with > the requirements of Freeswitch. Such as not having a MMU. So there doesn't > appear to be much effort porting Freeswitch to IP04 as a result. > > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. > > Are there any other good alternatives? > > Thanks. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From frank at carmickle.com Wed Dec 9 19:50:06 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 9 Dec 2009 22:50:06 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <20091210035006.GN31924@base.carmickle.com> On Thu, Dec 10, Brian May wrote: > Hello, > > I asked this question on my local linux user group mailing list, and got the > recommendation to ask here. > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > http://www.rowetel.com/ucasterisk/ip04.html > > It works well most of the time, however there are some bugs that do, under > circumstances lead to less then desirable behaviour (such as on some occasions > which I don't fully understand sometimes the remote system fails to generate > any audio packets when there is no audio - almost like silence suppression was > supported by the remote system - and asterisk fails to generate any audio > packets in return; on another slower computer running the same SIP software and > on the same network everything works fine; as far as I can tell the software - > twinkle - doesn't even support silence suppression). > > I suspect at least some - if not all - of the issues I have encountered may be > resolved with Freeswitch, however I don't really want to replace my small, > energy efficient, embedded system, with a large, power hungry computer system. > Overkill. > > An added complication is I need at least 1 analogue port to connect to the > Australian based telephone line (2 ports exchange ports and 1 extension port > would be ideal but not essiential). > > Unfortunately, I have been told that the IP04 hardware isn't compatable with > the requirements of Freeswitch. Such as not having a MMU. So there doesn't > appear to be much effort porting Freeswitch to IP04 as a result. > > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. > > Are there any other good alternatives? A board with an atom 330 on it would probably do the trick for you. There are a few made by Intel and Supermicro that look pretty nice. There were some other people on the list looking to use them. Maybe we can get a report from someone. --FC From jason at jasonjgw.net Wed Dec 9 19:55:29 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 10 Dec 2009 14:55:29 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <20091210035529.GA23724@jdc.jasonjgw.net> Brian May wrote: > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. Have a look at http://www.yawarra.com.au/ Some of their hardware (notably the Soekris Engineering boards: http://www.soekris.com/) has a PCI slot. Disclaimer: in principle this should work well with FreeSWITCH, but I haven't tested it as I don't own the hardware yet. From mouncifbb at gmail.com Wed Dec 9 20:06:02 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Wed, 9 Dec 2009 23:06:02 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> References: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> Message-ID: how big does need to get before it rotates, what's the size exactly? also how do I do it through dialplan via javascript? On Fri, Dec 4, 2009 at 6:48 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > set rotate-on-hup to false in the cdr_csv config file > then it will only rotate when the file gets too big > > and also you can get a cdr with > > session.generateXmlCdr() and dig out what you need or get it from > variables but it will not be nearly as reliable as using the C ones because > you need low level access to make sure you write to the disk properly from > many threads etc. > > > On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane wrote: > >> is it possible to run a javascript at the end of dialplan to generate >> cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file >> on machine reboots or shutdown signals. >> javascript or LUA for preferences? >> >> thank you >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/700ffd8d/attachment-0001.html From brian at microcomaustralia.com.au Wed Dec 9 20:53:32 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 15:53:32 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <4B207ECC.1020405@microcomaustralia.com.au> Kristian Kielhofner wrote: > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > Ok, I found this: . It looks like room for a full height card. 4 network adaptors for a Freeswitch box. Hmmm. Suspect I would only find use for one ;-) Lack of OpenZAP support might be an issue, I assume that would be required to connect to an onboard analogue port... I assume I could just install Debian or another distribution instead though. Does this require a hard disk drive to boot Linux? I am guessing that compact flash could be used instead. Alternatively, if I used an external ATA, what is a good one to use? I think Jason has already made a suggestion, if so I have forgotten. I guess I get nervous going down this approach because it will add to the latency, but then again it won't use so much CPU power either, and the Digium cards send a lot of time-critical interrupts. > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: I am curious, how do you install ISOs onto a box like the net5501? I don't see any provision for CD-ROM drives. -- Brian May From brian at microcomaustralia.com.au Wed Dec 9 21:21:20 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 16:21:20 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210035529.GA23724@jdc.jasonjgw.net> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035529.GA23724@jdc.jasonjgw.net> Message-ID: <4B208550.40104@microcomaustralia.com.au> Jason White wrote: > Have a look at http://www.yawarra.com.au/ > Ok, found the net5501: http://www.yawarra.com.au/hw-net5501.php And here it is assembled for you: http://www.yawarra.com.au/product.php?productCode=HW-NT55 I am not quite sure on one aspect, for extensions to work the TDM400P card requires a IDE style power connector that provides 12V, 5V, etc. Presumably this would be possible somehow with the net5501, because those voltages would be required for a HDD which seems to be supported. Anyone know what are the "Pigtail" and "DIN rail clips" options? -- Brian May From mike at jerris.com Wed Dec 9 23:18:00 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 02:18:00 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <3DB09BC7-3829-43B5-93A0-A1580768C7B0@jerris.com> I think I fixed the spandsp cross compile issues tonight, but I suspect there is a good chance that I broke other builds in the process. I also did a bunch of work to make the OS X Snow Leopard build cleaner today. Testing would be much appreciated on both. Mike On Dec 9, 2009, at 10:47 PM, Kristian Kielhofner wrote: > Brian, > > I have been making efforts to fully support FreeSWITCH in AstLinux. > Our primary targets are low powered x86 boards like the Soekris and > Alix. x86, powerful enough, cheap enough (as low as $100), and about > 12 watts. Not bad. > > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: > > http://mirror.astlinux.org/freeswitch/daily/ > > Let me know what you think. > > On Wed, Dec 9, 2009 at 7:55 PM, Brian May > wrote: >> Hello, >> >> I asked this question on my local linux user group mailing list, and got the >> recommendation to ask here. >> >> Anyway, at the moment I am running Asterisk on an IP04 embedded system. >> http://www.rowetel.com/ucasterisk/ip04.html >> >> It works well most of the time, however there are some bugs that do, under >> circumstances lead to less then desirable behaviour (such as on some occasions >> which I don't fully understand sometimes the remote system fails to generate >> any audio packets when there is no audio - almost like silence suppression was >> supported by the remote system - and asterisk fails to generate any audio >> packets in return; on another slower computer running the same SIP software and >> on the same network everything works fine; as far as I can tell the software - >> twinkle - doesn't even support silence suppression). >> >> I suspect at least some - if not all - of the issues I have encountered may be >> resolved with Freeswitch, however I don't really want to replace my small, >> energy efficient, embedded system, with a large, power hungry computer system. >> Overkill. >> >> An added complication is I need at least 1 analogue port to connect to the >> Australian based telephone line (2 ports exchange ports and 1 extension port >> would be ideal but not essiential). >> >> Unfortunately, I have been told that the IP04 hardware isn't compatable with >> the requirements of Freeswitch. Such as not having a MMU. So there doesn't >> appear to be much effort porting Freeswitch to IP04 as a result. >> >> I do have a spare TDM400p card, although as it is full height, suspect this >> isn't going to help. >> >> Are there any other good alternatives? >> >> Thanks. >> -- >> Brian May >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tculjaga at gmail.com Thu Dec 10 00:05:56 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 10 Dec 2009 09:05:56 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <65d96fc80912100005p5be13c65p50431025d882034a@mail.gmail.com> Kristian, from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is the maximum simultaneous calls that this box can handle of course using g729 codec? I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7 simultaneous calls. To be honest, i didnt run AstLinux on alix i used voyage instead but anyhow... this seems to be the limit. what i'm looking for it an appliance to run 2-16 FXS on it.... any suggestion? T. On Thu, Dec 10, 2009 at 4:47 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Brian, > > I have been making efforts to fully support FreeSWITCH in AstLinux. > Our primary targets are low powered x86 boards like the Soekris and > Alix. x86, powerful enough, cheap enough (as low as $100), and about > 12 watts. Not bad. > > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: > > http://mirror.astlinux.org/freeswitch/daily/ > > Let me know what you think. > > On Wed, Dec 9, 2009 at 7:55 PM, Brian May > wrote: > > Hello, > > > > I asked this question on my local linux user group mailing list, and got > the > > recommendation to ask here. > > > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > > http://www.rowetel.com/ucasterisk/ip04.html > > > > It works well most of the time, however there are some bugs that do, > under > > circumstances lead to less then desirable behaviour (such as on some > occasions > > which I don't fully understand sometimes the remote system fails to > generate > > any audio packets when there is no audio - almost like silence > suppression was > > supported by the remote system - and asterisk fails to generate any audio > > packets in return; on another slower computer running the same SIP > software and > > on the same network everything works fine; as far as I can tell the > software - > > twinkle - doesn't even support silence suppression). > > > > I suspect at least some - if not all - of the issues I have encountered > may be > > resolved with Freeswitch, however I don't really want to replace my > small, > > energy efficient, embedded system, with a large, power hungry computer > system. > > Overkill. > > > > An added complication is I need at least 1 analogue port to connect to > the > > Australian based telephone line (2 ports exchange ports and 1 extension > port > > would be ideal but not essiential). > > > > Unfortunately, I have been told that the IP04 hardware isn't compatable > with > > the requirements of Freeswitch. Such as not having a MMU. So there > doesn't > > appear to be much effort porting Freeswitch to IP04 as a result. > > > > I do have a spare TDM400p card, although as it is full height, suspect > this > > isn't going to help. > > > > Are there any other good alternatives? > > > > Thanks. > > -- > > Brian May > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/f722c55c/attachment.html From yehavi.bourvine at gmail.com Thu Dec 10 01:11:51 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 10 Dec 2009 11:11:51 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: An intermediate report: *Audiocodes*: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. *VegaStream Europa-50*: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: 2009/12/4 Yehavi Bourvine > I'll report when I am done. > > So far I've enabled only SRTP and both support it. > > __Yehavi: > > 2009/12/4 Mark Campbell-Smith > >> Thanks Yehavi, >> >> I would be very interested to find out how your test goes... can you >> report back after you have tested it? >> >> Thanks! >> >> On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine >> wrote: >> > Hello, >> > >> > I have AudioCodes MP and Vega ATA adapters. They both support SRTP; >> they >> > should support TLS also (will try it next week; up to now I preffered to >> not >> > use TLS so I can sniff the traffic and debug things). >> > >> > Regards, __Yehavi: >> > >> > 2009/12/4 Mark Campbell-Smith >> >> >> >> Cheers Gabriel.. thanks for the information. >> >> >> >> I'll look at the Mediatrix ATA's as an alternative - has anyone had >> >> experience with those and TLS/SRTP? >> >> >> >> >> >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: >> >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are >> the >> >> > Grandstream and Mediatrix devices (although I've never tried either >> >> > one with FreeSWITCH). >> >> > >> >> > I've personally never had any good experience with the Grandstream >> >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally >> >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they >> >> > support it (so says their marketing material and docs). >> >> > >> >> > I'd see if Cisco has any plans to add support for it to the ATAs. >> Next >> >> > time I see our Cisco SE, I'll try to poke him about it. >> >> > >> >> > Gabe >> >> > >> >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith >> >> > wrote: >> >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key >> exchange >> >> >> to appropriately support SRTP and FreeSWITCH >> >> >> >> >> >> I'll check with Cisco regarding their implementation then and try to >> >> >> find out when/if they will support standard SRTP encryption. >> >> >> >> >> >> >> >> >> So, back to my origianal question then. Are there any ATA's that >> >> >> support TLS AND SRTP with FreeSwitch? >> >> >> >> >> >> >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri >> wrote: >> >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do >> their >> >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on >> >> >>> upgrading the firmware to ever support SDES on the ATAs. They added >> >> >>> support for SDES to their IP Phones about 1 year ago, but nothing >> has >> >> >>> happened with the ATAs as of yet. >> >> >>> >> >> >>> Gabe >> >> >>> >> >> >>> >> >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> >> >>> wrote: >> >> >>>> Hi All, >> >> >>>> >> >> >>>> I managed to borrow a SPA3102 with the latest firmware and have >> got >> >> >>>> it >> >> >>>> to register using TLS, but I am still struggling with SRTP. Has >> >> >>>> anyone managed to get SRTP working with the Linksys devices and if >> >> >>>> so, >> >> >>>> can they direct me on how to do this. >> >> >>>> >> >> >>>> I have generated a mini-certificates and SRTP Private Key using >> the >> >> >>>> gen-mc tool found at >> >> >>>> >> >> >>>> >> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 >> . >> >> >>>> However, when ever I initiate a call from the SPA, I can see that >> >> >>>> the >> >> >>>> call is not encrypted. >> >> >>>> >> >> >>>> Help appreciated. >> >> >>>> >> >> >>>> Thanks! >> >> >>>> >> >> >>>> >> >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> >> >>>>> Check out the Linksys SPA2102 >> >> >>>>> >> >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> >> >>>>> wrote: >> >> >>>>>> >> >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> >> >>>>>> Grandstream HandyTone 503. But, again according to the wiki, >> that >> >> >>>>>> doesn't seem to behave to well with TLS ... >> >> >>>>>> >> >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White < >> jason at jasonjgw.net> >> >> >>>>>> wrote: >> >> >>>>>> > Mark Campbell-Smith wrote: >> >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? >> >> >>>>>> > >> >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, >> since >> >> >>>>>> > the >> >> >>>>>> > keys >> >> >>>>>> > would then be transmitted in the clear and therefore amenable >> to >> >> >>>>>> > interception. >> >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order >> to >> >> >>>>>> > be >> >> >>>>>> > secure. >> >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it >> works >> >> >>>>>> > entirely >> >> >>>>>> > in >> >> >>>>>> > RTP. >> >> >>>>>> > >> >> >>>>>> > I would be rather surprised were a hardware manufacturer to >> >> >>>>>> > implement >> >> >>>>>> > SRTP >> >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen >> >> >>>>>> > often in >> >> >>>>>> > this >> >> >>>>>> > forum that some manufacturers are really clueless... >> >> >>>>>> > >> >> >>>>>> > >> >> >>>>>> > _______________________________________________ >> >> >>>>>> > FreeSWITCH-users mailing list >> >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org >> >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>>> > >> >> >>>>>> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> > http://www.freeswitch.org >> >> >>>>>> > >> >> >>>>>> >> >> >>>>>> _______________________________________________ >> >> >>>>>> FreeSWITCH-users mailing list >> >> >>>>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>>> >> >> >>>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> http://www.freeswitch.org >> >> >>>>> >> >> >>>>> >> >> >>>>> _______________________________________________ >> >> >>>>> FreeSWITCH-users mailing list >> >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>> >> >> >>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >> >> >>>>> >> >> >>>>> >> >> >>>> >> >> >>>> _______________________________________________ >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> >> >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>> http://www.freeswitch.org >> >> >>>> >> >> >>> >> >> >>> _______________________________________________ >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> _______________________________________________ >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/0046d3cb/attachment-0001.html From asterisk at dotr.com Thu Dec 10 02:13:16 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Thu, 10 Dec 2009 10:13:16 +0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: Sometime next week I hopefully am going to start a document that follows my progress in setting up a FS system from scratch, with all the pitfalls and successes. A kinds of "warts and all" story. Alongside this "blog" (for want of a better word) I will also then document the steps needed to get it working (a howto guide, effectively). I am a long time * user (2004), so my mindset is kind of skewed - but perhaps that would be beneficial for other * users looking at implementing FS. Most of our config and dialplan is generated by using res_config_curl, and we use things like call listening, conferencing, parking and queues. We do use queues in a slightly odd manner (we add 1 agent, and call a local channel). When this channels is called, we use curl to get our application to return the most appropriate agent to actually call). We also use * as a power dialler, making upwards of 400,000 call attempts per month. Not massive, but not tiny either. Hopefully, this will be of use to both FS and * users. What would be great is that if other people follow my progress, and make suggestions as and when I hit a brick wall :) What would be best for this ? A blog ? Or a wiki page ? Julian 2009/12/9 Brian West : > That is what is nice about our community I'm more than willing to answer the > questions if you document them... as are many others in the core team...we > just have a lot to do and I think the best repayment is documentation! ;) > /b > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > > On Thu, Dec 10, 2009 at 11:07 AM, Brian West wrote: > > Visit the friday meetings and we can help if you document it. ?;) > > > I would be willing to lend a hand with the documentation but I know so > little (a complete freeswitch noob). For example I was trying to > figure out how to tell if an extension was set up "show dialplan in > asterisk". ?I could not find this anywhere. If I find out I would be > happy to add it to the rosetta stone. > > I am currently working on getting outbound socket working. Once I get > it going I would be happy to add it to the relevant section of the > wiki (in this case ruby). > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From codecomplete at free.fr Thu Dec 10 03:40:11 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 03:40:11 -0800 (PST) Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> Message-ID: <26725860.post@talk.nabble.com> No publisher, although uploading and selling books (deadtree or online) is easy with companies like www.lulu.com I was just thinking of some way to learn FS gradually and effectively. The frequent problem with wiki's, is that the quality of articles is uneven and they don't have a good layout. But then, writing documentation is hard and time-consuming :-/ -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p26725860.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 03:45:09 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 03:45:09 -0800 (PST) Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210035006.GN31924@base.carmickle.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> Message-ID: <26725918.post@talk.nabble.com> Frank Carmickle wrote: > A board with an atom 330 on it would probably do the trick for you. There > are a few made by Intel and Supermicro that look pretty nice. There were > some other people on the list looking to use them. Maybe we can get a > report from someone. Intel came up with the D945GSEJT, which is totally fanless and has an embedded DC/DC, so all you have to add is an external AC/DC power brick, some RAM, a PCI riser to save space, and either a hard-disk or a CompactFlash + IDE adaptor. I'm thinking of building one with a Digium-compatible PCI card. www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm -- View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mcampbellsmith at gmail.com Thu Dec 10 04:13:07 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 10 Dec 2009 23:13:07 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <26725918.post@talk.nabble.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> Message-ID: <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> I use a dectop by Data Evolution... Its cheap at ~$100. I have it running debian lenny and FS... works well for me. http://www.dataevolution.com/dectop%20info%202.htm http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/ On Thu, Dec 10, 2009 at 10:45 PM, Fred-145 wrote: > > > Frank Carmickle wrote: >> A board with an atom 330 on it would probably do the trick for you. ?There >> are a few made by Intel and Supermicro that look pretty nice. ?There were >> some other people on the list looking to use them. ?Maybe we can get a >> report from someone. > > Intel came up with the D945GSEJT, which is totally fanless and has an > embedded DC/DC, so all you have to add is an external AC/DC power brick, > some RAM, a PCI riser to save space, and either a hard-disk or a > CompactFlash + IDE adaptor. I'm thinking of building one with a > Digium-compatible PCI card. > > www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm > -- > View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From codecomplete at free.fr Thu Dec 10 04:57:21 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 04:57:21 -0800 (PST) Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> Message-ID: <26726783.post@talk.nabble.com> Mark Campbell-Smith wrote: > I use a dectop by Data Evolution... Its cheap at ~$100. I have it > running debian lenny and FS... works well for me. Thanks for the tip, although this type of box doesn't have a PCI slot, so the only way to connect FS to the PSTN is through a VoIP provider (or a Linksys 3102, with the usual, possible echo issues). In the same vein as the decTop : http://en.wikipedia.org/wiki/SheevaPlug Since FS can be compiled to ARM, Freeswitch might be able to run on this device: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux -- View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26726783.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 05:40:03 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 05:40:03 -0800 (PST) Subject: [Freeswitch-users] [vars.xml] default_password=1234? Message-ID: <26727371.post@talk.nabble.com> Hello I'm going through the various XML files, and noticed this first line in vars.xml. What is this password used for? Thank you -- View this message in context: http://old.nabble.com/-vars.xml--default_password%3D1234--tp26727371p26727371.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 10 05:45:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 07:45:28 -0600 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: I have confirmed it works with Polycom, Snom and a few others .... polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > An intermediate report: > > Audiocodes: TLS works only on outgoing requests, incoming ones are > ignored. I am waiting for Audiocodes' help in order to debug it. > SRTP: worked when no TLS is active. When TLS is active the call is > disconnected when the remote party answers. Still debugging it. > > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > how to enable TLS from the WEB interface. > > Regards, __Yehavi: From brian at freeswitch.org Thu Dec 10 05:47:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 07:47:28 -0600 Subject: [Freeswitch-users] [vars.xml] default_password=1234? In-Reply-To: <26727371.post@talk.nabble.com> References: <26727371.post@talk.nabble.com> Message-ID: please look in conf/directory/default/*.xml /b On Dec 10, 2009, at 7:40 AM, Fred-145 wrote: > > Hello > > I'm going through the various XML files, and noticed this first line > in > vars.xml. > > > > What is this password used for? > > Thank you From codecomplete at free.fr Thu Dec 10 06:04:38 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 06:04:38 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? Message-ID: <26727762.post@talk.nabble.com> Hello I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP activity" option unchecked, ran FreeSwitch, and used eg. Shields Up (www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened... which SU says no. Does it mean that... - by default, FS doesn't use STUN - or my modem doesn't support STUN, and I must either enable UPnP or map ports (UDP5080 and some UDP ports for RTP/RTPC) statically? Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26727762.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 06:09:17 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 06:09:17 -0800 (PST) Subject: [Freeswitch-users] [vars.xml] default_password=1234? In-Reply-To: References: <26727371.post@talk.nabble.com> Message-ID: <26727835.post@talk.nabble.com> Ah, makes sense: conf/directory/default/1000.xml: Thanks for the tip. -- View this message in context: http://old.nabble.com/-vars.xml--default_password%3D1234--tp26727371p26727835.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Dec 10 06:26:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 10 Dec 2009 15:26:17 +0100 Subject: [Freeswitch-users] Invite local number into a conference - codec problem Message-ID: <4B210509.4000004@gmx.net> Hello, I try to invite a user into a conference by loopback/255 8000 Conference 255 is the user, I invite the user via loopback as that way I can also invite external numbers. It processes the user's local dialplan correctly (as if the user was normally dialled), however it only offers L16 codec, so the Phone fails. I can see no codec negociation on the debug console. If I call the phone from another phone, then codec negociation is taking place. If I invite an external PSTN user into the conference then codecs are set correctly (L16+PCMA+PCMU etc) Is there a way to explicitely set the codec for the conference? is not set is, still commented in the internal profile. In vars.conf.xml only only PCMA and PCMU are set. Best regards Peter From frank at carmickle.com Thu Dec 10 06:26:58 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 10 Dec 2009 09:26:58 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <26725918.post@talk.nabble.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> Message-ID: <20091210142658.GP31924@base.carmickle.com> On Thu, Dec 10, Fred-145 wrote: > > > Frank Carmickle wrote: > > A board with an atom 330 on it would probably do the trick for you. There > > are a few made by Intel and Supermicro that look pretty nice. There were > > some other people on the list looking to use them. Maybe we can get a > > report from someone. > > Intel came up with the D945GSEJT, which is totally fanless and has an > embedded DC/DC, so all you have to add is an external AC/DC power brick, > some RAM, a PCI riser to save space, and either a hard-disk or a > CompactFlash + IDE adaptor. I'm thinking of building one with a > Digium-compatible PCI card. > > www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm The 330 boards are a little more power hungry but you get a dual core 64 bit processor. As far as I'm concerned the performance increase is well worth the extra money. You still well below the power consumption of any other 64 bit dual core machines. http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 --FC From rupa at rupa.com Thu Dec 10 06:33:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 10 Dec 2009 08:33:58 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26727762.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> Message-ID: STUN is not a way to open ports in a manner in which sheilds up would detect. http://en.wikipedia.org/wiki/STUN UPNP is what you want if you want to open ports. STUN is just a method for figuring out how to do nat traversal. STUN "method" is initiated by the process on the inside of the firewall at connection establish time. It will do no good for the listen case which is what you are checking for. If you want to use stun, then you need to forward the listen ports manually (5060, 5080 - UDP and TCP). On Thu, Dec 10, 2009 at 8:04 AM, Fred-145 wrote: > > Hello > > I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP > activity" option unchecked, ran FreeSwitch, and used eg. Shields Up > (www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened... > which SU says no. > > Does it mean that... > - by default, FS doesn't use STUN > - or my modem doesn't support STUN, and I must either enable UPnP or map > ports (UDP5080 and some UDP ports for RTP/RTPC) statically? > > Thank you. > -- > View this message in context: > http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26727762.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/670cff3f/attachment.html From kristian.kielhofner at gmail.com Thu Dec 10 08:12:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 10 Dec 2009 11:12:07 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210142658.GP31924@base.carmickle.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <20091210142658.GP31924@base.carmickle.com> Message-ID: <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle wrote: > > The 330 boards are a little more power hungry but you get a dual core 64 bit processor. ?As far as I'm concerned the performance increase is well worth the extra money. ?You still well below the power consumption of any other 64 bit dual core machines. > > http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 > > --FC > While these are low power when compared to traditional desktop/server systems, they're not what I would consider to be embedded. The CPU requires a fan (embedded no-no) and between the chipset and CPU they draw several times more power than a traditional embedded system. The ALIX and Soekris boards run with 12 watt power supplies (12v, 1 amp). The Atom 330 alone can draw 8 watts. This is still impressive for a processor of this class but it's not what I would consider to be embedded, yet... I think of embedded systems like this: Blackfin - Very low power, good performance (especially for DSP), very difficult porting (usually) ARM/MIPS - Very low power, decent performance depending on application, mild difficulty in porting X86 (Geode, etc) - Pretty low power, decent performance, relative ease in "porting" (often none) Everything else - You should probably call it an "appliance", not an embedded system With the correct target application and design ARM and Geode systems can provide more than enough CPU power for many, many practical applications. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Thu Dec 10 08:35:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 10:35:22 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: <191c3a030912100835p1cba7c5m5abca73d2ecfcc58@mail.gmail.com> Don't worry. I was an asterisk developer/volunteer in 2003. I still managed to figure it out. ;) On Thu, Dec 10, 2009 at 4:13 AM, Julian Lyndon-Smith wrote: > Sometime next week I hopefully am going to start a document that > follows my progress in setting up a FS system from scratch, with all > the pitfalls and successes. A kinds of "warts and all" story. > Alongside this "blog" (for want of a better word) I will also then > document the steps needed to get it working (a howto guide, > effectively). > > I am a long time * user (2004), so my mindset is kind of skewed - but > perhaps that would be beneficial for other * users looking at > implementing FS. > > Most of our config and dialplan is generated by using res_config_curl, > and we use things like call listening, conferencing, parking and > queues. We do use queues in a slightly odd manner (we add 1 agent, and > call a local channel). When this channels is called, we use curl to > get our application to return the most appropriate agent to actually > call). > > We also use * as a power dialler, making upwards of 400,000 call > attempts per month. Not massive, but not tiny either. > > Hopefully, this will be of use to both FS and * users. What would be > great is that if other people follow my progress, and make suggestions > as and when I hit a brick wall :) > > What would be best for this ? A blog ? Or a wiki page ? > > Julian > > 2009/12/9 Brian West : > > That is what is nice about our community I'm more than willing to answer > the > > questions if you document them... as are many others in the core > team...we > > just have a lot to do and I think the best repayment is documentation! ;) > > /b > > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > > > > On Thu, Dec 10, 2009 at 11:07 AM, Brian West > wrote: > > > > Visit the friday meetings and we can help if you document it. ;) > > > > > > I would be willing to lend a hand with the documentation but I know so > > little (a complete freeswitch noob). For example I was trying to > > figure out how to tell if an extension was set up "show dialplan in > > asterisk". I could not find this anywhere. If I find out I would be > > happy to add it to the rosetta stone. > > > > I am currently working on getting outbound socket working. Once I get > > it going I would be happy to add it to the relevant section of the > > wiki (in this case ruby). > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/18fbd093/attachment.html From abeka at greatiam.com Thu Dec 10 08:42:14 2009 From: abeka at greatiam.com (Otis) Date: Thu, 10 Dec 2009 16:42:14 +0000 Subject: [Freeswitch-users] Routing calls to Another FS server Message-ID: <4B2124E6.6010507@greatiam.com> I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set as gateways and register with each other. I wanted all users on FS1 to dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as 331001 on FS1. I have a dialplan as follows in .../dialplan/default/ callweaver.xml > > > > > I have also used I have also used the line in place of without any joy. I am getting error Not - found from the client. I am registered as 1001 on FS1. Please how do I make all users use this dial plan and may I know which version of all those stated above is right. All are in the ...dialplan/default directory. called callweaver.xml Should it have a particular name either than the gateway name ? Thanks for your time once again From anthony.minessale at gmail.com Thu Dec 10 08:53:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 10:53:44 -0600 Subject: [Freeswitch-users] Invite local number into a conference - codec problem In-Reply-To: <4B210509.4000004@gmx.net> References: <4B210509.4000004@gmx.net> Message-ID: <191c3a030912100853j3eea291bwd10d3e82b43e6dd8@mail.gmail.com> set absolute_codec_string to whatever codec you want to offer in the {} on the bridge string On Thu, Dec 10, 2009 at 8:26 AM, Peter P GMX wrote: > Hello, > > I try to invite a user into a conference by > loopback/255 8000 Conference > 255 is the user, I invite the user via loopback as that way I can also > invite external numbers. > > It processes the user's local dialplan correctly (as if the user was > normally dialled), however it only offers L16 codec, so the Phone fails. > I can see no codec negociation on the debug console. > If I call the phone from another phone, then codec negociation is taking > place. > If I invite an external PSTN user into the conference then codecs are > set correctly (L16+PCMA+PCMU etc) > > Is there a way to explicitely set the codec for the conference? > > is not set is, > still commented in the internal profile. > In vars.conf.xml only only PCMA and PCMU are set. > > Best regards > Peter > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/d6211aea/attachment-0001.html From abeka at greatiam.com Thu Dec 10 09:03:40 2009 From: abeka at greatiam.com (Otis) Date: Thu, 10 Dec 2009 17:03:40 +0000 Subject: [Freeswitch-users] Routing calls to Another FS server In-Reply-To: <4B2124E6.6010507@greatiam.com> References: <4B2124E6.6010507@greatiam.com> Message-ID: <4B2129EC.6000607@greatiam.com> Otis wrote: >
I have 2 > FS servers FS1 (aka medion) and FS3 (callweaver). These are set as > gateways and register with each other. I wanted all users on FS1 to > dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed > as 331001 on FS1. > > I have a dialplan as follows in .../dialplan/default/ callweaver.xml > >> >> >> >> >> > > I have also used > > > > data="sofia/profilename/$1 at 192.168.1.110"/> > > > > I have also used the line data="sofia/gateway/outbound.callweaver/$1"/> > in place of data="sofia/profilename/$1 at 192.168.1.110"/> > without any joy. > > I am getting error Not - found from the client. I am registered as > 1001 on FS1. > > Please how do I make all users use this dial plan and may I know > which version of all those stated above is right. All are in the > ...dialplan/default directory. called callweaver.xml > > Should it have a particular name either than the gateway name ? > > Thanks for your time once again > > > >
> Sorry I forgot to add that where it says profilename I have *callweaver* which is the profile name of the gateway in /conf/sip_profiles/external/callweaver.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/032209e8/attachment.html From aep.lists at it46.se Thu Dec 10 09:07:30 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 10 Dec 2009 18:07:30 +0100 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR Message-ID: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> Hi, I am currently creating IVR using the functions provided in the XML dialplan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr Using functions like this I can play files, etc. I wonder what is the smartest way to monitor (as in big brother) the options selected by the user: I assume that I can include an entry of the type: and include in foo.js the code to track the selection. But I wonder if this is the best approach /aep -- Stopping junk mailers is good for the environment From codecomplete at free.fr Thu Dec 10 09:21:43 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 09:21:43 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: References: <26727762.post@talk.nabble.com> Message-ID: <26731188.post@talk.nabble.com> Thanks for the clarification. So it's either UPnP or STUN/port-mapping. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26731188.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mailinglist at fribert.dk Thu Dec 10 09:27:19 2009 From: mailinglist at fribert.dk (mailinglist) Date: Thu, 10 Dec 2009 18:27:19 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <110796.60596.qm@web56401.mail.re3.yahoo.com> References: <4B2030A1020000E10000031D@mail.fribert.dk> <110796.60596.qm@web56401.mail.re3.yahoo.com> Message-ID: <4B213D87020000E100000322@mail.fribert.dk> Hi Mark The extensions are located under directory/default, and they look like this: As I understand them, the context set there, is the right one? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/57ee336e/attachment-0001.html From msc at freeswitch.org Thu Dec 10 09:51:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 09:51:55 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: <87f2f3b90912100951h74f3391ay91b354937417b741@mail.gmail.com> On Thu, Dec 10, 2009 at 2:13 AM, Julian Lyndon-Smith wrote: > Sometime next week I hopefully am going to start a document that > follows my progress in setting up a FS system from scratch, with all > the pitfalls and successes. A kinds of "warts and all" story. > Alongside this "blog" (for want of a better word) I will also then > document the steps needed to get it working (a howto guide, > effectively). > > I am a long time * user (2004), so my mindset is kind of skewed - but > perhaps that would be beneficial for other * users looking at > implementing FS. > > Most of our config and dialplan is generated by using res_config_curl, > and we use things like call listening, conferencing, parking and > queues. We do use queues in a slightly odd manner (we add 1 agent, and > call a local channel). When this channels is called, we use curl to > get our application to return the most appropriate agent to actually > call). > > We also use * as a power dialler, making upwards of 400,000 call > attempts per month. Not massive, but not tiny either. > > Hopefully, this will be of use to both FS and * users. What would be > great is that if other people follow my progress, and make suggestions > as and when I hit a brick wall :) > > What would be best for this ? A blog ? Or a wiki page ? > Julian, First off, welcome to the FreeSWITCH fold! Thank you for your willingness not only to try things but to document them as well. I like the idea of a blog to tell the story. After you're done then you might consider a wiki page that is more of "just the facts, ma'am" on how you translated Asterisk configs to a FreeSWITCH setup. Feel free to ask myself or anyone here or on IRC for assistance. -MC (IRC: mercutioviz) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/a7acf7b1/attachment.html From msc at freeswitch.org Thu Dec 10 09:53:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 09:53:16 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26725860.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> Message-ID: <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> On Thu, Dec 10, 2009 at 3:40 AM, Fred-145 wrote: > > No publisher, although uploading and selling books (deadtree or online) is > easy with companies like www.lulu.com > > I was just thinking of some way to learn FS gradually and effectively. The > frequent problem with wiki's, is that the quality of articles is uneven and > they don't have a good layout. But then, writing documentation is hard and > time-consuming :-/ > Amen, brothah! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/e639b372/attachment.html From msc at freeswitch.org Thu Dec 10 10:13:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:13:29 -0800 Subject: [Freeswitch-users] Routing calls to Another FS server In-Reply-To: <4B2124E6.6010507@greatiam.com> References: <4B2124E6.6010507@greatiam.com> Message-ID: <87f2f3b90912101013n339e7931nc50ba91328cdcb4b@mail.gmail.com> On Thu, Dec 10, 2009 at 8:42 AM, Otis wrote: > I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set > as gateways and register with each other. I wanted all users on FS1 to > dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as > 331001 on FS1. > > I have a dialplan as follows in .../dialplan/default/ callweaver.xml > > > > > > > > > > > > > I have also used > > > > > > > > I have also used the line data="sofia/gateway/outbound.callweaver/$1"/> > in place of data="sofia/profilename/$1 at 192.168.1.110"/> > without any joy. > > I am getting error Not - found from the client. I am registered as > 1001 on FS1. > Otis, I would recommend that you turn on debugging and capture the console output. Put it in a pastebin and then respond to this thread with a link to the pastebin URL. FYI, here's a handy page that will give you some nice pointers on how to gather information when you're doing troubleshooting: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has detailed instructions on getting console output, dumping to pastebin, etc. If you want to get really fancy you can also try Tony's handy-dandy debug Perl script: libs/esl/perl/logger.pl under the FS source directory. Run the script, make your call, then hit ctrl-C. The script will upload the log to pastebin for you and tell you the URL. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/8bb3e199/attachment.html From msc at freeswitch.org Thu Dec 10 10:15:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:15:56 -0800 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR In-Reply-To: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> References: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> Message-ID: <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero wrote: > > Hi, > > I am currently creating IVR using the functions provided in the XML > dialplan > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr > > Using functions like this > param="$${base_dir}/1255549537_Welcome.wav"/> > I can play files, etc. > > I wonder what is the smartest way to monitor (as in big brother) the > options selected by the user: > > I assume that I can include an entry of the type: > > and include in foo.js the code to track the selection. > > But I wonder if this is the best approach > > /aep > > Are you trying to do some sort of live monitoring as it happens (i.e. while the call is live) or do you just want a record of the digits they pressed? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/1ee57f3d/attachment.html From brian at freeswitch.org Thu Dec 10 10:17:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 12:17:06 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> Message-ID: Use the BKW method... three to four word sentences to describe what to do... its very poetic! Or is that haiku? /b On Dec 10, 2009, at 11:53 AM, Michael Collins wrote: > > I was just thinking of some way to learn FS gradually and > effectively. The > frequent problem with wiki's, is that the quality of articles is > uneven and > they don't have a good layout. But then, writing documentation is > hard and > time-consuming :-/ > > Amen, brothah! :) > -MC From mike at jerris.com Thu Dec 10 10:22:13 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 13:22:13 -0500 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26731188.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> Message-ID: <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> we also support natpmp and static ip setting. Mike On Dec 10, 2009, at 12:21 PM, Fred-145 wrote: > > Thanks for the clarification. So it's either UPnP or STUN/port-mapping. From msc at freeswitch.org Thu Dec 10 10:26:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:26:52 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> Message-ID: <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> On Thu, Dec 10, 2009 at 10:17 AM, Brian West wrote: > Use the BKW method... three to four word sentences to describe what to > do... its very poetic! Or is that haiku? > > /b > Update to latest Did you type make current yet? Tony hates build skew -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/12623dcc/attachment.html From shaun_clark at hotmail.com Thu Dec 10 10:26:39 2009 From: shaun_clark at hotmail.com (Shaun Clark) Date: Thu, 10 Dec 2009 10:26:39 -0800 Subject: [Freeswitch-users] Route Non-Call Data to Agent Through Queue Message-ID: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> I have an application where I would like to route both calls and other requests through the same queue to the same agents, for example the same agent might take a call and then right after that take a chat. But, the chat server we use is separate from our phone system. What I would like to do is basically route some text, i.e. "new chat chat_id_goes_here" through to the agent. Is this possible with FreeSwitch? The idea being the soft-phone would receive this text and we would write code to catch this message do the appropriate action on our CRM. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/bce228c0/attachment-0001.html From aep.lists at it46.se Thu Dec 10 11:24:12 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 10 Dec 2009 20:24:12 +0100 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR In-Reply-To: <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> References: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> Message-ID: <770de553a9869a67b2db3a1bc5fae765.squirrel@correo.nodo50.org> I want to trigger CUSTOM events via ESL "as they navigate inside" of the IVR. The XML IVRs are generated from a GUI. The CUSTOM events need to carry - what IVR the user is navigating - what option has been selected - ideally how long they stayed listening (this can be calculated) - and when they hang the phone /aep -- Stopping junk mailers is good for the environment > On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero > wrote: > >> >> Hi, >> >> I am currently creating IVR using the functions provided in the XML >> dialplan >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr >> >> Using functions like this >> > param="$${base_dir}/1255549537_Welcome.wav"/> >> I can play files, etc. >> >> I wonder what is the smartest way to monitor (as in big brother) the >> options selected by the user: >> >> I assume that I can include an entry of the type: >> >> and include in foo.js the code to track the selection. >> >> But I wonder if this is the best approach >> >> /aep >> >> Are you trying to do some sort of live monitoring as it happens (i.e. >> while > the call is live) or do you just want a record of the digits they pressed? > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tculjaga at gmail.com Thu Dec 10 12:53:51 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 10 Dec 2009 21:53:51 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <20091210142658.GP31924@base.carmickle.com> <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> Message-ID: <65d96fc80912101253o410ff73dv213b4c6509f882ae@mail.gmail.com> ok, but how much smultaneous calls did you get on an alix board using astlinux... for istnace, this is the question? T. On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle > wrote: > > > > The 330 boards are a little more power hungry but you get a dual core 64 > bit processor. As far as I'm concerned the performance increase is well > worth the extra money. You still well below the power consumption of any > other 64 bit dual core machines. > > > > http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 > > > > --FC > > > > While these are low power when compared to traditional > desktop/server systems, they're not what I would consider to be > embedded. The CPU requires a fan (embedded no-no) and between the > chipset and CPU they draw several times more power than a traditional > embedded system. The ALIX and Soekris boards run with 12 watt power > supplies (12v, 1 amp). The Atom 330 alone can draw 8 watts. This is > still impressive for a processor of this class but it's not what I > would consider to be embedded, yet... > > I think of embedded systems like this: > > Blackfin - Very low power, good performance (especially for DSP), very > difficult porting (usually) > ARM/MIPS - Very low power, decent performance depending on > application, mild difficulty in porting > X86 (Geode, etc) - Pretty low power, decent performance, relative ease > in "porting" (often none) > Everything else - You should probably call it an "appliance", not an > embedded system > > With the correct target application and design ARM and Geode systems > can provide more than enough CPU power for many, many practical > applications. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/b0c3804f/attachment.html From dave at 3c.co.uk Thu Dec 10 13:11:20 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 10 Dec 2009 14:11:20 -0700 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> Message-ID: <1260479480.12078.18.camel@local.freepabx.com> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: > Update to latest > Did you type make current yet? > Tony hates build skew Brilliant. Michael Collins-san Shrinks all usual advice Into one Haiku. --Dave From msc at freeswitch.org Thu Dec 10 13:19:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 13:19:32 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <1260479480.12078.18.camel@local.freepabx.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> Message-ID: <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> On Thu, Dec 10, 2009 at 1:11 PM, David Knell wrote: > > On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: > > > Update to latest > > Did you type make current yet? > > Tony hates build skew > > Brilliant. > > Michael Collins-san > Shrinks all usual advice > Into one Haiku. > > --Dave > I love the wiki The docs are disorganized I hate the wiki -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/530c6846/attachment.html From asterisk at dotr.com Thu Dec 10 14:16:54 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Thu, 10 Dec 2009 22:16:54 +0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> Message-ID: Ok. The journey begins. http://makingfs.blogspot.com/ Don't know if you want to add this link to the website or wiki. Julian 2009/12/10 Michael Collins : > > > On Thu, Dec 10, 2009 at 1:11 PM, David Knell wrote: >> >> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: >> >> > Update to latest >> > Did you type make current yet? >> > Tony hates build skew >> >> Brilliant. >> >> Michael Collins-san >> Shrinks all usual advice >> Into one Haiku. >> >> --Dave > > I love the wiki > The docs are disorganized > I hate the wiki > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From andrew at hijacked.us Thu Dec 10 14:32:58 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 10 Dec 2009 17:32:58 -0500 Subject: [Freeswitch-users] Route Non-Call Data to Agent Through Queue In-Reply-To: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> References: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> Message-ID: <20091210223258.GA14106@hijacked.us> On Thu, Dec 10, 2009 at 10:26:39AM -0800, Shaun Clark wrote: > I have an application where I would like to route both calls and other > requests through the same queue to the same agents, for example the same > agent might take a call and then right after that take a chat. But, the chat > server we use is separate from our phone system. > > What I would like to do is basically route some text, i.e. "new chat > chat_id_goes_here" through to the agent. Is this possible with FreeSwitch? > The idea being the soft-phone would receive this text and we would write > code to catch this message do the appropriate action on our CRM. Thanks! I did this by writing my own external queueing (in erlang) and simply parking the calls in FS and adding them to my external queue (along with emails, voicemails, etc). With asterisk I added a fake call to the queue with some channel variables that referenced the external data I was really putting in the queue and I listened for the 'BRIDGE' event on the AMI and sent the agent the external data then. I'm not sure mod_fifo needs to be a universal queue - but maybe you could do what you want via api_after_bridge and uuid_chat or something crazy? You'd have to script whatever soft-phone you're using to be smart about that though. Andrew From msc at freeswitch.org Thu Dec 10 14:34:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 14:34:44 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> Message-ID: <87f2f3b90912101434j4f4f9d9bm67a5f789b0cadd@mail.gmail.com> On Thu, Dec 10, 2009 at 2:16 PM, Julian Lyndon-Smith wrote: > Ok. The journey begins. > > http://makingfs.blogspot.com/ > > Don't know if you want to add this link to the website or wiki. > > Julian > > Excellent work! Thanks, -MC Asterisk deadlocked Why does it suck so badly? Use FreeSWITCH instead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/3c1658c8/attachment.html From ken at ksac.com Thu Dec 10 16:30:03 2009 From: ken at ksac.com (Kendall Stauffer) Date: Thu, 10 Dec 2009 16:30:03 -0800 Subject: [Freeswitch-users] windows pre compiled asr Message-ID: I downloaded yesterdays latest pre compiled and seems to works great, but I get invalid Asr module when trying to run pizza app. It seemed to come pre configured with pocketsphynx, anything I should know before I spend a boat load of time on it? Rest seems real good,. thatks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/7100ed86/attachment-0001.html From ken at ksac.com Thu Dec 10 16:37:02 2009 From: ken at ksac.com (Kendall Stauffer) Date: Thu, 10 Dec 2009 16:37:02 -0800 Subject: [Freeswitch-users] never mind Message-ID: Sorry, I now see it wasn't loaded, so must not come with the pre compiled. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/71caf1a8/attachment.html From msc at freeswitch.org Thu Dec 10 16:49:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 16:49:23 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Dec 11 Message-ID: <87f2f3b90912101649x500c0bc9kfaefc770fbf25a8@mail.gmail.com> FYI, Here's the agenda for tomorrow's conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11 Please be ready to join at 11AM CST! :) Don't forget to bring your agenda items, questions, and a willingness to help out with our various janitor projects. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/256f225a/attachment.html From brian at microcomaustralia.com.au Thu Dec 10 17:57:24 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Fri, 11 Dec 2009 12:57:24 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <4B207ECC.1020405@microcomaustralia.com.au> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> Message-ID: <20091211015724.GC14547@sys11.in.vpac.org> On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: > Lack of OpenZAP support might be an issue, I assume that would be > required to connect to an onboard analogue port... I assume I could just > install Debian or another distribution instead though. This is another distribution I found: http://linux.voyage.hk/ It comes with Asterisk out of the box, although I suspect it wouldn't be too hard to get Freeswitch working instead. -- Brian May From mike at jerris.com Thu Dec 10 18:20:39 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 21:20:39 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211015724.GC14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: As a note, we are pretty aggressive about making sure all this stuff works right out of svn without any patches so it should be easy to port freeswitch to most platforms now. Mike On Dec 10, 2009, at 8:57 PM, Brian May wrote: > On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: >> Lack of OpenZAP support might be an issue, I assume that would be >> required to connect to an onboard analogue port... I assume I could just >> install Debian or another distribution instead though. > > This is another distribution I found: > > http://linux.voyage.hk/ > > It comes with Asterisk out of the box, although I suspect it > wouldn't be too hard to get Freeswitch working instead. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at microcomaustralia.com.au Thu Dec 10 18:42:28 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Fri, 11 Dec 2009 13:42:28 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: <20091211024228.GE14547@sys11.in.vpac.org> On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote: > As a note, we are pretty aggressive about making sure all this stuff works > right out of svn without any patches so it should be easy to port freeswitch > to most platforms now. Thats good to hear. I am guessing this means I should use a recent version. I see there is an Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should be able to build from the source. Anyway I sent an email to Yawarra to ask them if the net5501 computer is compatible with the TDM400 cards. There is something about a kit for the dual rack mount computer for the TDM400, which would be good if I had a rack, and somewhere to put a rack. So presumably this means it should work for the non-rack mount system too. -- Brian May From JCasale at activenetwerx.com Thu Dec 10 19:30:27 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Fri, 11 Dec 2009 03:30:27 +0000 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211024228.GE14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> <20091211024228.GE14547@sys11.in.vpac.org> Message-ID: >Anyway I sent an email to Yawarra to ask them if the net5501 computer > is >compatible with the TDM400 cards. It is, people have been doing this for a while w/ astlinux: http://www.mail-archive.com/astlinux-users at lists.sourceforge.net/msg03048.html jlc From mcampbellsmith at gmail.com Thu Dec 10 20:10:26 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 11 Dec 2009 15:10:26 +1100 Subject: [Freeswitch-users] Passing user variables to mod_voicemail Message-ID: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Hi! My voip provider provides a SOAP interface to be able to send SMS's, so after a voicemail is left, I want to execute a 'send sms' script. I don't want a separate statement in the dialplan after the voicemail statement because I only want to send sms's when a voicemail is actually left. The way I was going to do this was to modify the mailer-app to point to a shell script and modify the mailer-app-args to include some user defined variables (in conf/directory/default/*.xml). The shell script would do the following: emailvm.sh #$1 $2 $3 = smsaccount smspassword textmessage tee /tmp/vmmail | /usr/sbin/sendmail -t exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log However, if I uncomment the last line, I never see the user variables being passed to the shell script. The email is sucessfully sent, but the sms script doesnt work. If fact, the output of log.log is (for example): -f 1001 at 192.168.1.120 email_address at domain.com Any ideas if it is possible to pass user variables via mod_voicemail in this way? Thanks From anthony.minessale at gmail.com Thu Dec 10 21:28:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 23:28:17 -0600 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Message-ID: <191c3a030912102128n76e35268l7aa057fd1ff205d7@mail.gmail.com> That wont work. I'm not sure if there is a way, I cant think of one off the top of my head. On Thu, Dec 10, 2009 at 10:10 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > My voip provider provides a SOAP interface to be able to send SMS's, > so after a voicemail is left, I want to execute a 'send sms' script. > I don't want a separate statement in the dialplan after the voicemail > statement because I only want to send sms's when a voicemail is > actually left. > > The way I was going to do this was to modify the mailer-app to point > to a shell script and modify the mailer-app-args to include some user > defined variables (in conf/directory/default/*.xml). > > value="/usr/local/freeswitch/scripts/emailvm.sh"/> > > > The shell script would do the following: > > emailvm.sh > > #$1 $2 $3 = smsaccount smspassword textmessage > tee /tmp/vmmail | /usr/sbin/sendmail -t > exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 > #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log > > However, if I uncomment the last line, I never see the user variables > being passed to the shell script. The email is sucessfully sent, but > the sms script doesnt work. If fact, the output of log.log is (for > example): > > -f 1001 at 192.168.1.120 email_address at domain.com > > Any ideas if it is possible to pass user variables via mod_voicemail > in this way? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/289b8826/attachment.html From codecomplete at free.fr Fri Dec 11 00:46:00 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 00:46:00 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> Message-ID: <26740589.post@talk.nabble.com> Michael Jerris wrote: > we also support natpmp and static ip setting. What is "static ip setting"? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? So, in the case where the FS server is located in a private network, these are the ways to open up the ports it needs to allow remote SIP users to connect to it: - UPnP and NAT-PMP (FS asks the router for its public IP, and negotiates opening the required UDP ports dynamically) - STUN (to get the public IP address from a remote STUN server) + port-mapping (to permanently open required UDP ports on NAT firewall) - possibly this fourth solution above -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26740589.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Fri Dec 11 00:59:03 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 11 Dec 2009 09:59:03 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211015724.GC14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: <65d96fc80912110059u5b2b7a5h389421716563aa54@mail.gmail.com> voyage linux is a stripped debian and i was using it on an alix board some time ago... Asterisk was compiling on that without any issue. I beleive FS will do the same. T. On Fri, Dec 11, 2009 at 2:57 AM, Brian May wrote: > On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: > > Lack of OpenZAP support might be an issue, I assume that would be > > required to connect to an onboard analogue port... I assume I could just > > install Debian or another distribution instead though. > > This is another distribution I found: > > http://linux.voyage.hk/ > > It comes with Asterisk out of the box, although I suspect it > wouldn't be too hard to get Freeswitch working instead. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/969cfd3e/attachment.html From mctch at yahoo.com Fri Dec 11 01:44:12 2009 From: mctch at yahoo.com (Mark Crane) Date: Fri, 11 Dec 2009 01:44:12 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B213D87020000E100000322@mail.fribert.dk> Message-ID: <117186.49301.qm@web56406.mail.re3.yahoo.com> What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark ? The extensions are located under directory/default, and they look like this: ? ??? ????? ????? ????? ????? ????? ????? ??? ??? ????? ????? ????? ????? ????? ??? ? ? ??? ????? ????? ????? ????? ????? ????? ??? ??? ????? ????? ????? ??? ? As I understand them, the context set there, is the right one? ? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. ????? Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! ? It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? ? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml ? ? The default.xml looks like this ( I haven't changed it): ? ? ? ??? ????? ????? ? ????? ??? ? ??? ??? ????? ????? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ????? ? ????? ????? ????? ????? ? ? ? ????? ? ????? ? ? ? ????? ??? ? ??? ??? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ??? ??? ?? ??? ??? ????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ? ? ??? ??? ????? ? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ??? ????? ? ????? ??? ? ??? ?????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ??????? ??????? ? ????? ??? ? ??? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ????? ? ????? ??? ??? ??? ????? ????? ????? ? ? ????? ??? ? ??? ??? ????? ????? ????? ?]]> ? ????? ??? ??? ??? ??? ????? ????? ????? ? ? ?????? ??? ? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ??????? ??????? ????? ??? ? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ? ??? ????? ????? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ??? ??? ? ? ??? ??? ????? ??? ? ? ??? ??? ??? ? ??? ? ? ? ? Then I have under default dir: musimidk.xml ?? ?????? ?? and 9000_recordings.xml ?? ?????? ?? ? ? Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/c6baee33/attachment-0001.html From Russell.Mosemann at cune.org Fri Dec 11 04:33:45 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Fri, 11 Dec 2009 06:33:45 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26740589.post@talk.nabble.com> References: <26727762.post@talk.nabble.com><26731188.post@talk.nabble.com><0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> Message-ID: Fred-145 wrote: > What is "static ip setting"? Telling FS what the public IP is? If that's > what it is, what about the UDP ports that must be open to allow incoming > connections? Yes, static IP setting puts the (non-changing) IP addresses in the FS configuration. The ports must be manually opened/forwarded in the firewall. -- Russell Mosemann From pjintheusa at gmail.com Fri Dec 11 05:52:21 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 11 Dec 2009 08:52:21 -0500 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Message-ID: <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> Hi - sorry to go off topic - but we are looking for Voip supplier with SMS capability. Would you mind telling me which Voip supplier you use? On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > My voip provider provides a SOAP interface to be able to send SMS's, > so after a voicemail is left, I want to execute a 'send sms' script. > I don't want a separate statement in the dialplan after the voicemail > statement because I only want to send sms's when a voicemail is > actually left. > > The way I was going to do this was to modify the mailer-app to point > to a shell script and modify the mailer-app-args to include some user > defined variables (in conf/directory/default/*.xml). > > value="/usr/local/freeswitch/scripts/emailvm.sh"/> > > > The shell script would do the following: > > emailvm.sh > > #$1 $2 $3 = smsaccount smspassword textmessage > tee /tmp/vmmail | /usr/sbin/sendmail -t > exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 > #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log > > However, if I uncomment the last line, I never see the user variables > being passed to the shell script. The email is sucessfully sent, but > the sms script doesnt work. If fact, the output of log.log is (for > example): > > -f 1001 at 192.168.1.120 email_address at domain.com > > Any ideas if it is possible to pass user variables via mod_voicemail > in this way? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/6fc32154/attachment.html From mcampbellsmith at gmail.com Fri Dec 11 06:28:03 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 12 Dec 2009 01:28:03 +1100 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> Message-ID: <33c87fa30912110628r14923948q11909e9175f768bc@mail.gmail.com> Pennytel.com On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones wrote: > Hi - sorry to go off topic - but we are looking for Voip supplier with SMS > capability. Would you mind telling me which Voip supplier you use? > > On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> My voip provider provides a SOAP interface to be able to send SMS's, >> so after a voicemail is left, I want to execute a 'send sms' script. >> I don't want a separate statement in the dialplan after the voicemail >> statement because I only want to send sms's when a voicemail is >> actually left. >> >> The way I was going to do this was to modify the mailer-app to point >> to a shell script and modify the mailer-app-args to include some user >> defined variables (in conf/directory/default/*.xml). >> >> ? ?> value="/usr/local/freeswitch/scripts/emailvm.sh"/> >> ? ? >> >> The shell script would do the following: >> >> emailvm.sh >> >> #$1 $2 $3 = smsaccount smspassword textmessage >> tee /tmp/vmmail | /usr/sbin/sendmail -t >> exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 >> #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log >> >> However, if I uncomment the last line, I never see the user variables >> being passed to the shell script. ?The email is sucessfully sent, but >> the sms script doesnt work. ?If fact, the output of log.log is (for >> example): >> >> -f 1001 at 192.168.1.120 email_address at domain.com >> >> Any ideas if it is possible to pass user variables via mod_voicemail >> in this way? >> >> Thanks >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ken at ksac.com Fri Dec 11 07:47:55 2009 From: ken at ksac.com (Kendall Stauffer) Date: Fri, 11 Dec 2009 07:47:55 -0800 Subject: [Freeswitch-users] Still cant find it Message-ID: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/021854da/attachment.html From chris.chen2004 at gmail.com Fri Dec 11 08:06:09 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 11:06:09 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks Message-ID: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. I tested with both Polycom IP650 and Bria 2.5.4, compared against port audio and googletalk endpoints in the same network. all SIP end points (Polycom and Bria) behind NAT but in the same subnet 192.168.0, I tried to change the settings below: in /conf/sip_profiles/internal.xml using different combinations of either enabling or disabling them. the results are all the same, the audios on sip endpoints always got cut about 31 seconds, no issues at all with either port audio or gtalk, Could anyone point me to the right direction for the sofia_sip profile setup? Your helps are greatly appreciated Thanks, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/76e54d74/attachment.html From brian at freeswitch.org Fri Dec 11 08:08:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 10:08:47 -0600 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: > Ok, So I have looked around a lot now, think I have read everything > carefully, and don?t see an answer to my questions anywhere, but > apologize if it is already somewhere. > SO I need the sphinx and tts modules, and don?t see their src on > the site with the freeswitch stuff. > Do I just download from CMU? Any certain versions? > Would be nice if somebody already compiled for windows > I am very impressed with freeswitch, and thank you for your > efforts!!! > > _______________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/91a1d3bc/attachment.html From jeff at jefflenk.com Fri Dec 11 08:23:05 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 11 Dec 2009 08:23:05 -0800 (PST) Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: <1260548585714-4152195.post@n2.nabble.com> The source tarballs are downloaded by the vs2008 project files when you build the solution Kendall Stauffer wrote: > > Ok, So I have looked around a lot now, think I have read everything > carefully, and don't see an answer to my questions anywhere, but apologize > if it is already somewhere. > SO I need the sphinx and tts modules, and don't see their src on the > site with the freeswitch stuff. > Do I just download from CMU? Any certain versions? > Would be nice if somebody already compiled for windows > I am very impressed with freeswitch, and thank you for your efforts!!! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Still-cant-find-it-tp4152031p4152195.html Sent from the freeswitch-users mailing list archive at Nabble.com. From frank at carmickle.com Fri Dec 11 08:25:00 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 11 Dec 2009 11:25:00 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> Message-ID: <20091211162459.GR31924@base.carmickle.com> On Fri, Dec 11, Chris Chen wrote: > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC From ken at ksac.com Fri Dec 11 08:25:33 2009 From: ken at ksac.com (Kendall Stauffer) Date: Fri, 11 Dec 2009 08:25:33 -0800 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: Yes, I can do that , I don't see where I download the source, Sorry to bug you. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, December 11, 2009 11:09 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Still cant find it Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! _______________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/42877b15/attachment.html From chris.chen2004 at gmail.com Fri Dec 11 08:44:27 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 11:44:27 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <20091211162459.GR31924@base.carmickle.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> Message-ID: <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as "deny", that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > On Fri, Dec 11, Chris Chen wrote: > > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > > trunk 15905. > > Is this a change in behavior or is this the first time you've run > freeswitch? If this is your first time welcome aboard! Also if this is > your first time you've probably have some IPs aliased on your interface and > you still have stun enabled. This was the behavior I saw the first time I > ran it on a box with aliases on an interface. The stun server tells > freeswitch after some time that the IP is different then the one you've > assigned. This is just one possibility. If this isn't the case then we > will need to see sip traces on all of your profiles. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/89ffe944/attachment.html From carlos.talbot at gmail.com Fri Dec 11 09:04:06 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 11 Dec 2009 11:04:06 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: References: Message-ID: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13>ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: 1 unresolved externals regards, Carlos On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer wrote: > I downloaded yesterdays latest pre compiled and seems to works great, but > I get invalid Asr module when trying to run pizza app. > > It seemed to come pre configured with pocketsphynx, anything I should > know before I spend a boat load of time on it? > > Rest seems real good,. thatks!!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/23e8e033/attachment.html From msc at freeswitch.org Fri Dec 11 09:04:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Dec 2009 09:04:07 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting! Message-ID: <87f2f3b90912110904y716e85f9o6afa42a1960da70f@mail.gmail.com> Come one, come all! http://bit.ly/8KzHCZ Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/75833c93/attachment.html From brian at freeswitch.org Fri Dec 11 09:08:10 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 11:08:10 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> References: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> Message-ID: <28E7DE31-320E-47D7-86CD-40619CF37269@freeswitch.org> Thats being fixed today! ;) /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: > It hasn't been included as of late since I'm getting an unresolved > link error during the build. I'll need someone experienced in > pocketsphinx to assist with this issue: > > 13>ngram_search.obj : error LNK2001: unresolved external symbol > _ngram_model_flush > 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: > 1 unresolved externals > > regards, > > Carlos > From kristian.kielhofner at gmail.com Fri Dec 11 09:14:25 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 11 Dec 2009 12:14:25 -0500 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket Message-ID: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> Hello everyone, PB here: http://pastebin.freeswitch.org/11482 FS rev 15909. The relevant bits from the log are here (starting around line 135): # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 sofia/pjsip/nobody at 192.168.4.253 Command Execute bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj at 192.168.4.17) # EXECUTE sofia/pjsip/nobody at 192.168.4.253 bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj at 192.168.4.17) # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 0 = [originate_timeout=30] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 1 = [bypass_media=true] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 2 = [origination_caller_id_number=1111] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 3 = [origination_caller_id_name=Tara Ousley] bypass_media=true yet the SDP of the outgoing INVITE looks like this: # send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: # ------------------------------------------------------------------------ # INVITE sip:huttoj at 192.168.4.17 SIP/2.0 # Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr # Max-Forwards: 69 # From: "Tara Ousley" ;tag=0tU8SN9pvejNK # To: # Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 # CSeq: 124148250 INVITE # Contact: # User-Agent: Voalte Voice # Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY # Supported: timer, precondition, path, replaces # Allow-Events: talk, refer # Content-Type: application/sdp # Content-Disposition: session # Content-Length: 271 # X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 # Remote-Party-ID: "Tara Ousley" ;party=calling;screen=yes;privacy=off # # v=0 # o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 # s=FreeSWITCH # c=IN IP4 192.168.2.10 # t=0 0 # m=audio 25172 RTP/AVP 9 0 101 # a=rtpmap:9 G722/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=silenceSupp:off - - - - # a=ptime:20 192.168.2.10 is the address of my FS box... -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Fri Dec 11 09:31:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Dec 2009 11:31:26 -0600 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket In-Reply-To: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> References: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> Message-ID: <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> Hey, You can't set bypass_media=true in {} or it will not take effect unless that b leg itself becomes an a leg some day. you need to execute set on bypass_media=true on the leg before you call bridge to trigger it. Alternatively you could set {bypass_media_after_bridge=true} or set it on A leg as described above on either leg and it will do the bypass once the audio is flowing. On Fri, Dec 11, 2009 at 11:14 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > PB here: > > http://pastebin.freeswitch.org/11482 > > FS rev 15909. The relevant bits from the log are here (starting > around line 135): > > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 > sofia/pjsip/nobody at 192.168.4.253 Command Execute > > bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara > Ousley}sofia/voalte/huttoj at 192.168.4.17) > # > EXECUTE sofia/pjsip/nobody at 192.168.4.253 > > bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara > Ousley}sofia/voalte/huttoj at 192.168.4.17) > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 0 = [originate_timeout=30] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 1 = [bypass_media=true] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 2 = [origination_caller_id_number=1111] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 3 = [origination_caller_id_name=Tara Ousley] > > bypass_media=true yet the SDP of the outgoing INVITE looks like this: > > # > send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: > # > ------------------------------------------------------------------------ > # > INVITE sip:huttoj at 192.168.4.17 SIP/2.0 > # > Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr > # > Max-Forwards: 69 > # > From: "Tara Ousley" > >;tag=0tU8SN9pvejNK > # > To: > > # > Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 > # > CSeq: 124148250 INVITE > # > Contact: > # > User-Agent: Voalte Voice > # > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > # > Supported: timer, precondition, path, replaces > # > Allow-Events: talk, refer > # > Content-Type: application/sdp > # > Content-Disposition: session > # > Content-Length: 271 > # > X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 > # > Remote-Party-ID: "Tara Ousley" > > >;party=calling;screen=yes;privacy=off > # > > # > v=0 > # > o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 > # > s=FreeSWITCH > # > c=IN IP4 192.168.2.10 > # > t=0 0 > # > m=audio 25172 RTP/AVP 9 0 101 > # > a=rtpmap:9 G722/8000 > # > a=rtpmap:0 PCMU/8000 > # > a=rtpmap:101 telephone-event/8000 > # > a=fmtp:101 0-16 > # > a=silenceSupp:off - - - - > # > a=ptime:20 > > 192.168.2.10 is the address of my FS box... > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/35e93a9f/attachment-0001.html From zendel.fernandez at gmail.com Fri Dec 11 03:02:47 2009 From: zendel.fernandez at gmail.com (zendel fernandez) Date: Fri, 11 Dec 2009 16:32:47 +0530 Subject: [Freeswitch-users] gtalk dingaling G723 Message-ID: hi! Pls shed some light to the below dingaling/gtalk issue. ___________________________Call path________________________ Gtalk ---> Account setup in client.xml ---routed to public.xml ---> route to a external SIP gateway -----> sipUserA ___________________________Problem statement________________ The above call scenario is successfull using the *PMCU, PCMA* codecs in "dingaling.conf.xml". Both parties can hear each other. When I choose G723 the call disconnects after few seconds(2~). Both parties can hear some noise & thasts all. I see the call is bridged and then immediately unbridged. I assume G.723 is suppose to work here in passtru mode. Do I have to do any other configuration ohter than the ones listed below in order to get passthru working. ___________________________LOG important parts________________ (Full log found at the end) 2009-12-11 16:05:12.948700 [ERR] mod_g723_1.c:148 This codec is only usable in passthrough mode! 2009-12-11 16:05:12.948700 [DEBUG] switch_ivr_bridge.c:464 DingaLing/new ending bridge by request from write function 2009-12-11 16:05:12.948700 [DEBUG] switch_ivr_bridge.c:520 sofia/external/0094777915380 receive message [UNBRIDGE] ________________________Configuration XMLS______________________ ________________________dingaling.conf.xml_____________________ ______________________________public.xml__________________________ ______________________________client.xml___________________________ __________________________________Full Log__________________________ ____________________________________________________________________ ____________________________________________________________________ ____________________________________________________________________ freeswitch at internal> 2009-12-11 16:18:05.848176 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:353 Created Session 596627090 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [ISAC] id='103' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [IPCMWB] id='97' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [G723] id='4' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [EG711U] id='100' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [EG711A] id='101' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [PCMU] id='0' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [PCMA] id='8' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [CN] id='13' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [iLBC] id='102' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [red] id='117' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [audio/telephone-event] id='106' 2009-12-11 16:18:05.848176 [DEBUG] mod_dingaling.c:3028 Creating an identity for 596627090 GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB < GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB> 5555 2009-12-11 16:18:05.848176 [NOTICE] switch_channel.c:613 New Channel dingaling/5555 [d689728e-fcbd-433d-8b87-7c54c03d2825] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3056 Creating a session for 596627090 2009-12-11 16:18:05.849387 [NOTICE] switch_channel.c:611 Rename Channel dingaling/5555->DingaLing/new [d689728e-fcbd-433d-8b87-7c54c03d2825] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3060 (DingaLing/new) State Change CS_NEW -> CS_INIT 2009-12-11 16:18:05.849387 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3157 11 payloads 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload ISAC 103 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare ISAC 103/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload IPCMWB 97 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare IPCMWB 97/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload G723 4 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare G723 4/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3177 Choosing Payload index 0 G723 4 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:1066 Send Describe [G723 at 8000] 2009-12-11 16:18:05.849387 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_INIT 2009-12-11 16:18:05.849387 [DEBUG] switch_core_state_machine.c:338 (DingaLing/new) State INIT 2009-12-11 16:18:05.849387 [NOTICE] mod_dingaling.c:1093 Ring-Ready DingaLing/new! 2009-12-11 16:18:05.884166 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.884166 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:05.884166 [DEBUG] libdingaling.c:1406 Sending packet 300 (2 left) 2009-12-11 16:18:05.884166 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.983333 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.084116 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.184092 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.284068 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.384042 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.483334 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.583993 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.683965 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.783940 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.867919 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:06.867919 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:06.867919 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=Unokk7lKh84I8mEM password=At5OQzApw/ZrIrCo address=172.16.11.110 port=1563 pref=1.00 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:3227 1 candidates 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:3243 candidate 172.16.11.110:1563 FAIL ACL wan 2009-12-11 16:18:06.883915 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:06.883915 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.983330 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.013899 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=W3xgvc25vCXb5M/U password=7EcqsVB3S6J3I2+g address=172.16.7.28 port=1566 pref=1.00 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:3227 1 candidates 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:3243 candidate 172.16.7.28:1566 FAIL ACL wan 2009-12-11 16:18:07.083870 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.083870 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.119862 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.119862 [DEBUG] libdingaling.c:943 Cancel packet 300 2009-12-11 16:18:07.183848 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.183848 [DEBUG] libdingaling.c:1414 Discarding packet 300 2009-12-11 16:18:08.365552 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:08.365552 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:08.365552 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=relay protocol=udp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=19295 pref=0.50 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3227 2 candidates 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3243 candidate 172.16.7.28:1566 FAIL ACL wan 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3239 candidate 209.85.229.126:19295 PASS ACL wan 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3288 Acceptable Candidate 209.85.229.126:19295 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:976 Stun Lookup Local 172.16.11.211:22878 2009-12-11 16:18:15.496911 [INFO] mod_dingaling.c:984 Stun Success XX.XX.XX.XX:22878 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:998 Send Candidate XX.XX.XX.XX:22878 [goTRR95uTkRRMe3t] 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:856 Set Read Codec to G723 at 8000 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:871 Set Write Codec to G723 at 8000 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:884 SETUP RTP 172.16.11.211:22878 -> 209.85.229.126:19295 2009-12-11 16:18:15.496911 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 960 bytes per 120ms 2009-12-11 16:18:15.498911 [DEBUG] switch_rtp.c:2905 Activate VAD codec G723 120ms 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) State Change CS_INIT -> CS_ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:338 (DingaLing/new) State INIT going to sleep 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:341 (DingaLing/new) State ROUTING 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1198 DingaLing/new CHANNEL ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:78 DingaLing/new Standard ROUTING 2009-12-11 16:18:15.498911 [INFO] mod_dialplan_xml.c:408 Processing GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB->5555 in context public Dialplan: DingaLing/new parsing [public->public_did] continue=false Dialplan: DingaLing/new Regex (PASS) [public_did] caller_id_number( GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB) =~ /^([^@]+)/ break=never Dialplan: DingaLing/new Action set(effective_caller_id_number=GTALK_ANY_CLIENT) Dialplan: DingaLing/new Regex (PASS) [public_did] destination_number(5555) =~ /^(5555)$/ break=on-false Dialplan: DingaLing/new Action set(call_timeout=18) Dialplan: DingaLing/new Action set(continue_on_fail=true) Dialplan: DingaLing/new Action set(hangup_after_bridge=true) Dialplan: DingaLing/new Action bridge(sofia/gateway/sbc/sipUserA) Dialplan: DingaLing/new Action answer() Dialplan: DingaLing/new Action voicemail(default 172.16.11.211 1001) 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:122 (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:15.499911 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:341 (DingaLing/new) State ROUTING going to sleep 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:348 (DingaLing/new) State EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] mod_dingaling.c:1215 DingaLing/new CHANNEL EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:159 DingaLing/new Standard EXECUTE EXECUTE DingaLing/new set(effective_caller_id_number=GTALK_ANY_CLIENT) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [effective_caller_id_number]=[GTALK_ANY_CLIENT] EXECUTE DingaLing/new set(call_timeout=18) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [call_timeout]=[18] EXECUTE DingaLing/new set(continue_on_fail=true) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [continue_on_fail]=[true] EXECUTE DingaLing/new set(hangup_after_bridge=true) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [hangup_after_bridge]=[true] EXECUTE DingaLing/new bridge(sofia/gateway/sbc/sipUserA) 2009-12-11 16:18:15.500912 [NOTICE] switch_channel.c:613 New Channel sofia/external/sipUserA [cf250379-b2c9-439e-8295-766feef6cf74] 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:3142 (sofia/external/sipUserA) State Change CS_NEW -> CS_INIT 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_INIT 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:338 (sofia/external/sipUserA) State INIT 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:83 sofia/external/sipUserA SOFIA INIT 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:111 (sofia/external/sipUserA) State Change CS_INIT -> CS_ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:338 (sofia/external/sipUserA) State INIT going to sleep 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:341 (sofia/external/sipUserA) State ROUTING 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:132 sofia/external/sipUserA SOFIA ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_ivr_originate.c:66 (sofia/external/sipUserA) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:341 (sofia/external/sipUserA) State ROUTING going to sleep 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_CONSUME_MEDIA 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:360 (sofia/external/sipUserA) State CONSUME_MEDIA 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:360 (sofia/external/sipUserA) State CONSUME_MEDIA going to sleep 2009-12-11 16:18:15.501911 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [calling][0] 2009-12-11 16:18:15.583888 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:15.583888 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.583888 [DEBUG] libdingaling.c:1406 Sending packet 301 (2 left) 2009-12-11 16:18:15.583888 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:15.684864 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.784839 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.884815 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.984789 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.084764 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.184740 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.284715 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.384691 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.484666 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.584640 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.684617 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.784591 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.810584 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:16.810584 [DEBUG] libdingaling.c:943 Cancel packet 301 2009-12-11 16:18:16.884567 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.884567 [DEBUG] libdingaling.c:1414 Discarding packet 301 2009-12-11 16:18:17.795342 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.795342 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:17.795342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=local protocol=tcp username=pBOc+m0nq7uQngnq password=HzZY2+zb/u9oRH++ address=172.16.7.28 port=1569 pref=0.80 2009-12-11 16:18:17.795342 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:17.795342 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:17.798342 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=tcp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=19294 pref=0.50 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=local protocol=tcp username=EXe2XpIc76S9WxgB password=BKF7SUs/hc30Fplp address=172.16.11.110 port=1570 pref=0.80 2009-12-11 16:18:17.798342 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:17.798342 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:17.884322 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.884322 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:18.797094 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=ssltcp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=443 pref=0.50 2009-12-11 16:18:18.797094 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:18.797094 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:18.885072 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:21.529273 [INFO] sofia.c:506 Update Callee ID to "sipUserA" 2009-12-11 16:18:21.529273 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [proceeding][180] 2009-12-11 16:18:21.529273 [DEBUG] sofia.c:3738 Remote SDP: v=0 o=- 298002432 1260522569 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 15002 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=ptime:30 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[G723:4:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[PCMU:0:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[PCMA:8:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[GSM:3:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3353 Substituting codec G723 at 30i@8000h 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:2143 Set Codec sofia/external/sipUserA G723/8000 30 ms 240 samples 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:2381 AUDIO RTP [sofia/external/sipUserA] 172.16.11.211 port 32408 -> YY.YY.YY.YY port 15002 codec: 4 ms: 30 2009-12-11 16:18:21.530280 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 240 bytes per 30ms 2009-12-11 16:18:21.531273 [NOTICE] sofia_glue.c:2909 Pre-Answer sofia/external/sipUserA! 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:2020 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_originate.c:2886 DingaLing/new receive message [PROGRESS] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [NOTICE] switch_ivr_originate.c:2886 Pre-Answer DingaLing/new! 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_originate.c:2929 Originate Resulted in Success: [sofia/external/sipUserA] 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:182 sofia/external/sipUserA receive message [AUDIO_SYNC] 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:182 DingaLing/new receive message [AUDIO_SYNC] 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1032 sofia/external/sipUserA receive message [BRIDGE] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1039 DingaLing/new receive message [BRIDGE] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1083 (sofia/external/sipUserA) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] switch_core_state_machine.c:351 (sofia/external/sipUserA) State EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] mod_sofia.c:455 SOFIA LOOPBACK 2009-12-11 16:18:21.548268 [DEBUG] switch_rtp.c:2004 Correct ip/port confirmed. 2009-12-11 16:18:23.513922 [INFO] sofia.c:506 Update Callee ID to "sipUserA" 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [completing][200] 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3735 Duplicate SDP v=0 o=- 298002432 1260522569 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 15002 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=ptime:30 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [ready][200] 2009-12-11 16:18:23.513922 [DEBUG] switch_channel.c:2133 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.513922 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.513922 [NOTICE] sofia.c:4219 Channel [sofia/external/sipUserA] has been answered 2009-12-11 16:18:23.513922 [DEBUG] switch_channel.c:182 sofia/external/sipUserA receive message [AUDIO_SYNC] 2009-12-11 16:18:23.587904 [DEBUG] switch_ivr_bridge.c:394 DingaLing/new receive message [ANSWER] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.587904 [NOTICE] switch_ivr_bridge.c:394 Channel [DingaLing/new] has been answered 2009-12-11 16:18:23.587904 [DEBUG] switch_channel.c:182 DingaLing/new receive message [AUDIO_SYNC] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:706 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:706 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.617898 [DEBUG] switch_ivr_bridge.c:131 sofia/external/sipUserA receive message [DISPLAY] 2009-12-11 16:18:23.708875 [DEBUG] switch_ivr_bridge.c:131 DingaLing/new receive message [DISPLAY] 2009-12-11 16:18:25.568413 [DEBUG] switch_rtp.c:2004 Correct ip/port confirmed. 2009-12-11 16:18:25.568413 [ERR] mod_g723_1.c:148 This codec is only usable in passthrough mode! 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:464 DingaLing/new ending bridge by request from write function 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:520 sofia/external/sipUserA receive message [UNBRIDGE] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_session.c:645 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD DONE [sofia/external/sipUserA] 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:565 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.568413 [NOTICE] switch_ivr_bridge.c:617 Hangup sofia/external/sipUserA [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 16:18:25.568413 [DEBUG] switch_channel.c:1912 Send signal sofia/external/sipUserA [KILL] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:488 (sofia/external/sipUserA) State HANGUP 2009-12-11 16:18:25.568413 [DEBUG] mod_sofia.c:358 Channel sofia/external/sipUserA hanging up, cause: NORMAL_CLEARING 2009-12-11 16:18:25.568413 [DEBUG] mod_sofia.c:400 Sending BYE to sofia/external/sipUserA 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:46 sofia/external/sipUserA Standard HANGUP, cause: NORMAL_CLEARING 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:488 (sofia/external/sipUserA) State HANGUP going to sleep 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:351 (sofia/external/sipUserA) State EXCHANGE_MEDIA going to sleep 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_HANGUP 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:465 sofia/external/sipUserA handler already called, skipping state handler. 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:333 (sofia/external/sipUserA) State Change CS_HANGUP -> CS_REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:579 (sofia/external/sipUserA) State REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:53 sofia/external/sipUserA Standard REPORTING, cause: NORMAL_CLEARING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:579 (sofia/external/sipUserA) State REPORTING going to sleep 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:327 (sofia/external/sipUserA) State Change CS_REPORTING -> CS_DESTROY 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:1136 Session 2 (sofia/external/sipUserA) Locked, Waiting on external entities 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:520 DingaLing/new receive message [UNBRIDGE] 2009-12-11 16:18:25.628397 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD DONE [DingaLing/new] 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:565 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.628397 [NOTICE] switch_ivr_bridge.c:1179 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 16:18:25.628397 [DEBUG] switch_channel.c:1912 Send signal DingaLing/new [KILL] 2009-12-11 16:18:25.628397 [DEBUG] libdingaling.c:298 Destroyed Session 596627090 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:488 (DingaLing/new) State HANGUP 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1293 DingaLing/new CHANNEL HANGUP 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:46 DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:488 (DingaLing/new) State HANGUP going to sleep 2009-12-11 16:18:25.628397 [NOTICE] switch_core_session.c:1154 Session 2 (sofia/external/sipUserA) Ended 2009-12-11 16:18:25.628397 [NOTICE] switch_core_session.c:1156 Close Channel sofia/external/sipUserA [CS_DESTROY] 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:348 (DingaLing/new) State EXECUTE going to sleep 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:423 (sofia/external/sipUserA) Running State Change CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_HANGUP 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipUserA) State DESTROY 2009-12-11 16:18:25.629398 [DEBUG] mod_sofia.c:293 sofia/external/sipUserA SOFIA DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:465 DingaLing/new handler already called, skipping state handler. 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipUserA Standard DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:579 (DingaLing/new) State REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipUserA) State DESTROY going to sleep 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:53 DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:579 (DingaLing/new) State REPORTING going to sleep 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:327 (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:1136 Session 1 (DingaLing/new) Locked, Waiting on external entities 2009-12-11 16:18:25.629398 [NOTICE] switch_core_session.c:1154 Session 1 (DingaLing/new) Ended 2009-12-11 16:18:25.629398 [NOTICE] switch_core_session.c:1156 Close Channel DingaLing/new [CS_DESTROY] 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:423 (DingaLing/new) Running State Change CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (DingaLing/new) State DESTROY 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1231 NUKE RTP 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:60 DingaLing/new Standard DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (DingaLing/new) State DESTROY going to sleep 2009-12-11 16:18:25.683383 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.683383 [DEBUG] libdingaling.c:1406 Sending packet 302 (2 left) 2009-12-11 16:18:25.683383 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:25.783359 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.883332 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.983307 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.083280 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.183267 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.283231 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.383206 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.483181 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.583156 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.651139 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.651139 [DEBUG] libdingaling.c:943 Cancel packet 302 2009-12-11 16:18:26.660136 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.660136 [DEBUG] libdingaling.c:353 Created Session 596627090 2009-12-11 16:18:26.660136 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:26.660136 [DEBUG] mod_dingaling.c:2926 Session is already dead 2009-12-11 16:18:26.683132 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.683132 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.683132 [DEBUG] libdingaling.c:1414 Discarding packet 302 Regds. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/ac7d3bc6/attachment-0001.html From mike at jerris.com Fri Dec 11 09:51:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 12:51:13 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> Message-ID: <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly > I noticed that the acl automatically having 192.168.0.0 set as "deny", that's why I tried to changed the settings regarding nat acl and localnet acl. > > Chris > > > > On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > On Fri, Dec 11, Chris Chen wrote: > > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > > trunk 15905. > > Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/917eb7b4/attachment.html From mike at jerris.com Fri Dec 11 09:55:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 12:55:13 -0500 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote: > Yes, I can do that , I don?t see where I download the source, Sorry to bug you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/694aec32/attachment.html From brian at freeswitch.org Fri Dec 11 09:56:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 11:56:52 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> References: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> Message-ID: Can you confirm its fixed now? /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: > It hasn't been included as of late since I'm getting an unresolved > link error during the build. I'll need someone experienced in > pocketsphinx to assist with this issue: > > 13>ngram_search.obj : error LNK2001: unresolved external symbol > _ngram_model_flush > 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: > 1 unresolved externals > > regards, > > Carlos > > On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer > wrote: > I downloaded yesterdays latest pre compiled and seems to works > great, but I get invalid Asr module when trying to run pizza app. > > It seemed to come pre configured with pocketsphynx, anything I > should know before I spend a boat load of time on it? > > Rest seems real good,. thatks!!! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/2538794a/attachment.html From brian at freeswitch.org Fri Dec 11 10:03:51 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:03:51 -0600 Subject: [Freeswitch-users] gtalk dingaling G723 In-Reply-To: References: Message-ID: <958069D8-A221-4F8A-A155-430AFB141393@freeswitch.org> Can't use G723. /b On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote: > > hi! > > Pls shed some light to the below dingaling/gtalk issue. > From codecomplete at free.fr Fri Dec 11 10:09:09 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 10:09:09 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> Message-ID: <26748901.post@talk.nabble.com> One last question: Does someone know of a utility for Windows that can check that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal to write a small diagnostic by connecting to free firewall checkers to see if the relevant ports are open, but if it's already available... Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26748901.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Dec 11 10:13:21 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:13:21 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26748901.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> Message-ID: <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> FreeSWITCH on windows will already poke holes in the windows firewall using upnp. Just start FS and it works. Your outer nat is a larger issue... /b On Dec 11, 2009, at 12:09 PM, Fred-145 wrote: > > One last question: Does someone know of a utility for Windows that > can check > that a NAT router supports either UPnP or NAT-PMP? I guess it's no > big deal > to write a small diagnostic by connecting to free firewall checkers > to see > if the relevant ports are open, but if it's already available... > > Thank you. > -- From asterisk at dotr.com Fri Dec 11 10:19:39 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Fri, 11 Dec 2009 18:19:39 +0000 Subject: [Freeswitch-users] The Building Freeswitch blog Message-ID: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian From brian at freeswitch.org Fri Dec 11 10:24:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:24:38 -0600 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: <4A6BCF50-D8E1-4D8B-BED0-6616A5BAB8A2@freeswitch.org> well mod_alas.c is for the N800 Please open a jira. /b On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote: > Doing the building thing, seem to have come across a bug. > > Have a look at Part 2 of http://makingfs.blogspot.com/ > > If make crashes out, it states that it was successfully built ;) > > Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/52960711/attachment-0001.html From mike at jerris.com Fri Dec 11 10:34:45 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 13:34:45 -0500 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. Mike p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: > Doing the building thing, seem to have come across a bug. > > Have a look at Part 2 of http://makingfs.blogspot.com/ > > If make crashes out, it states that it was successfully built ;) > > Julian > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From asterisk at dotr.com Fri Dec 11 10:47:13 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Fri, 11 Dec 2009 18:47:13 +0000 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: Thanks Mike. I understand why you don't want all to be built. However, there are things that I would like - such as mod_java. However, that fails to compile, I presume because of some missing dependency or requirement. Is there any tool to tell me what is needed in order to build a module ? Julian 2009/12/11 Michael Jerris : > It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. ?The basics of this is that automake continues on to other subdirs if build in one subdir fails. > > Mike > > p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. ?This module is barely touched and I would not use it unless you have a good reason to. > > > On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: > >> Doing the building thing, seem to have come across a bug. >> >> Have a look at Part 2 of http://makingfs.blogspot.com/ >> >> If make crashes out, it states that it was successfully built ;) >> >> Julian >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Fri Dec 11 11:03:58 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:03:58 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> Message-ID: <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris wrote: > As i said multiple times on irc last night, we need to see debug logs with > sip trace to see what is going on. > > Mike > > On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > > Thanks Frank for sharing your experience. This is the behavior change just > starting within three days, maybe because of some code changes in mod_sofia > which I should change the settings accordingly > I noticed that the acl automatically having 192.168.0.0 set as "deny", > that's why I tried to changed the settings regarding nat acl and localnet > acl. > > Chris > > > > On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > >> On Fri, Dec 11, Chris Chen wrote: >> > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN >> > trunk 15905. >> >> Is this a change in behavior or is this the first time you've run >> freeswitch? If this is your first time welcome aboard! Also if this is >> your first time you've probably have some IPs aliased on your interface and >> you still have stun enabled. This was the behavior I saw the first time I >> ran it on a box with aliases on an interface. The stun server tells >> freeswitch after some time that the IP is different then the one you've >> assigned. This is just one possibility. If this isn't the case then we >> will need to see sip traces on all of your profiles. >> >> --FC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/432df45f/attachment.html From mrene_lists at avgs.ca Fri Dec 11 11:09:06 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 11 Dec 2009 14:09:06 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Message-ID: Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: > Hi Mike, the fs console log with sip trace on the internal profile > is attached in the pastebin below, > http://pastebin.freeswitch.org/11483 > > could you please take a look at it? > Thanks, > Chris > > On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris > wrote: > As i said multiple times on irc last night, we need to see debug > logs with sip trace to see what is going on. > > Mike > > On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > >> Thanks Frank for sharing your experience. This is the behavior >> change just starting within three days, maybe because of some code >> changes in mod_sofia which I should change the settings accordingly >> I noticed that the acl automatically having 192.168.0.0 set as >> "deny", that's why I tried to changed the settings regarding nat >> acl and localnet acl. >> >> Chris >> >> >> >> On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle > > wrote: >> On Fri, Dec 11, Chris Chen wrote: >> > Hi there, I have very strange behaviors for my SIP endpoints with >> FS SVN >> > trunk 15905. >> >> Is this a change in behavior or is this the first time you've run >> freeswitch? If this is your first time welcome aboard! Also if >> this is your first time you've probably have some IPs aliased on >> your interface and you still have stun enabled. This was the >> behavior I saw the first time I ran it on a box with aliases on an >> interface. The stun server tells freeswitch after some time that >> the IP is different then the one you've assigned. This is just one >> possibility. If this isn't the case then we will need to see sip >> traces on all of your profiles. >> >> --FC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/d467295a/attachment.html From mike at jerris.com Fri Dec 11 11:11:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 14:11:37 -0500 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: <09EB5512-D5EC-489C-9C47-9BB3A8F6A502@jerris.com> Probably the best list is: http://wiki.freeswitch.org/wiki/FreeSwitch_Dependencies Due to the fact that we allow you to change modules after configure there is no great way to have it error out when you don't have the right deps other than to just have the compile errors when you try to build. Its probably time for a tool like make menuconfig but we do not have that as of yet. Mike On Dec 11, 2009, at 1:47 PM, Julian Lyndon-Smith wrote: > Thanks Mike. I understand why you don't want all to be built. However, > there are things that I would like - such as mod_java. However, that > fails to compile, I presume because of some missing dependency or > requirement. Is there any tool to tell me what is needed in order to > build a module ? > > Julian > > 2009/12/11 Michael Jerris : >> It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. >> >> Mike >> >> p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. >> >> >> On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: >> >>> Doing the building thing, seem to have come across a bug. >>> >>> Have a look at Part 2 of http://makingfs.blogspot.com/ >>> >>> If make crashes out, it states that it was successfully built ;) >>> >>> Julian >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris.chen2004 at gmail.com Fri Dec 11 11:34:26 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:34:26 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Message-ID: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Thanks Mathieu, but I am on SVN r15912 now. Chris On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene wrote: > Its not sending to the right Contact: header in the 200 OK packet. This > was fixed in r15870, you have to update. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 11-Dec-09, at 2:03 PM, Chris Chen wrote: > > Hi Mike, the fs console log with sip trace on the internal profile is > attached in the pastebin below, > http://pastebin.freeswitch.org/11483 > > could you please take a look at it? > Thanks, > Chris > > On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris wrote: > >> As i said multiple times on irc last night, we need to see debug logs with >> sip trace to see what is going on. >> >> Mike >> >> On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: >> >> Thanks Frank for sharing your experience. This is the behavior change just >> starting within three days, maybe because of some code changes in mod_sofia >> which I should change the settings accordingly >> I noticed that the acl automatically having 192.168.0.0 set as "deny", >> that's why I tried to changed the settings regarding nat acl and localnet >> acl. >> >> Chris >> >> >> >> On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: >> >>> On Fri, Dec 11, Chris Chen wrote: >>> > Hi there, I have very strange behaviors for my SIP endpoints with FS >>> SVN >>> > trunk 15905. >>> >>> Is this a change in behavior or is this the first time you've run >>> freeswitch? If this is your first time welcome aboard! Also if this is >>> your first time you've probably have some IPs aliased on your interface and >>> you still have stun enabled. This was the behavior I saw the first time I >>> ran it on a box with aliases on an interface. The stun server tells >>> freeswitch after some time that the IP is different then the one you've >>> assigned. This is just one possibility. If this isn't the case then we >>> will need to see sip traces on all of your profiles. >>> >>> --FC >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/756d2adc/attachment.html From kristian.kielhofner at gmail.com Fri Dec 11 11:41:19 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 11 Dec 2009 14:41:19 -0500 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket In-Reply-To: <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> References: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> Message-ID: <2d9149cd0912111141r575be558s14d2002a9cee1d6e@mail.gmail.com> Thanks, that was it! On Fri, Dec 11, 2009 at 12:31 PM, Anthony Minessale wrote: > Hey, > > You can't set bypass_media=true in {} or it will not take effect unless that > b leg itself becomes an a leg some day. > you need to execute set on bypass_media=true on the leg before you call > bridge to trigger it. > > Alternatively you could set {bypass_media_after_bridge=true} or set it on A > leg as described above on either leg and it will do the bypass once the > audio is flowing. > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Fri Dec 11 11:45:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 13:45:23 -0600 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Message-ID: <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris From brian at freeswitch.org Fri Dec 11 11:56:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 13:56:49 -0600 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Message-ID: Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris From chris.chen2004 at gmail.com Fri Dec 11 11:57:49 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:57:49 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> Message-ID: <507898380912111157r56f88862ua498166904aed2b4@mail.gmail.com> Thanks Brian for your explanation, could we still keep the option to set the extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe many other routers have similar issue. Chris On Fri, Dec 11, 2009 at 2:45 PM, Brian West wrote: > You set the extrtp ip to an IP exactly.. this is the issue we are > fixing soon.. if you have natpmp or upnp set it to auto-nat and let it > figure it out. The issue is we have restored the behavior in 1.0.4 > that lies about the IP all the time... > > I'm going to commit a patch shortly that'll fix this. > > /b > > On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > > > Thanks Mathieu, but I am on SVN r15912 now. > > > > Chris > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/7f1c2a9d/attachment.html From bcxml at hotmail.com Fri Dec 11 15:11:31 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 15:11:31 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound Message-ID: <26752894.post@talk.nabble.com> I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass an incomming PSTN call to Speech Server. But I cannot get Freeswitch to dial out a call or transfer a call that is sent from Speech Server I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called voipms.xml which contains the following..(I have an account with voip.ms) And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called Outbound.xml which contains the following When my Speech Server application tries to get FreeSwitch to transfer to another number, the console shows the following 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/+1 9059183027 at 199.173.95.16:5060 to XML[%23904161234 at public] 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/interna l/2482578002 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing +14165551212 ->%23904161234 in context public 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/ internal/2482578002 at 127.0.0.1:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sof ia/internal/2482578002 at 127.0.0.1:5060 [CS_DESTROY] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup sofia /external/+19059183027 at 199.173.95.16:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/ external/+19059183027 at 199.173.95.16:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sof ia/external/+19059183027 at 199.173.95.16:5060 [CS_DESTROY] I really dont understand the line above that I have in Bold & Italic The number being transfered to was 4161234567...so I would have thought the line should read.. Processing +14165551212->4161234567 in context public Can anyone tell me what the "%2390" means and also any problems with my XML files that could be preventing the transfers from taking place Thanks Brian -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26752894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bcxml at hotmail.com Fri Dec 11 15:23:19 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 15:23:19 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound Message-ID: <26752894.post@talk.nabble.com> I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass an incomming PSTN call to Speech Server. But I cannot get Freeswitch to dial out a call or transfer a call that is sent from Speech Server I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called voipms.xml which contains the following..(I have an account with voip.ms) And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called Outbound.xml which contains the following When my Speech Server application tries to get FreeSwitch to transfer to another number, the console shows the following 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/+14052582456 at 111.222.333.444:5060 to XML[%23904161234 at public] 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/2484487788 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing +14052582456->%23904161234 in context public 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/internal/2484487788 at 127.0.0.1:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2484487788 at 127.0.0.1:5060 [CS_DESTROY] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/+14052582456 at 111.222.333.444:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/+14052582456 at 111.222.333.444:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/+14052582456 at 111.222.333.444:5060 [CS_DESTROY] I really dont understand the line above that I have in Bold & Italic The number being transfered to was 4161234567...so I would have thought the line should read.. "Processing +14052582456->4161234567 in context public " Can anyone tell me what the "%2390" means and also any problems with my XML files that could be preventing the transfers from taking place Thanks Brian -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26752894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Fri Dec 11 15:34:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Dec 2009 15:34:14 -0800 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <26752894.post@talk.nabble.com> References: <26752894.post@talk.nabble.com> Message-ID: <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> On Fri, Dec 11, 2009 at 3:11 PM, bcxml wrote: > > I am very new to Freeswitch so please accept my appologies if these > questions > seem to be trivial > > I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. > I > have been successful in getting Freeswitch to pass an incomming PSTN call > to > Speech Server. But I cannot get Freeswitch to dial out a call or transfer a > call that is sent from Speech Server > > I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called > voipms.xml which contains the following..(I have an account with voip.ms) > > > > > > > > > > > > And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called > Outbound.xml which contains the following > > > > data="effective_caller_id_number=12223334444"/> > > > > > When my Speech Server application tries to get FreeSwitch to transfer to > another number, the console shows the following > > > 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer > sofia/external/+1 > 9059183027 at 199.173.95.16:5060 to XML[%23904161234 at public] > 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup > sofia/interna > l/2482578002 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing > +14165551212 > ->%23904161234 in context public > This line is basically saying that you have a call coming from 4165551212 and it's looking for a destination number of %23904161234. The key here is that it is coming in the public context so you'll need to handle the routing in conf/dialplan/public.xml What should this call be doing once it comes in to FS? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/c85d280d/attachment.html From codecomplete at free.fr Fri Dec 11 15:38:19 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 15:38:19 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> Message-ID: <26753168.post@talk.nabble.com> Right, when talking about NAT firewall, I meant the outer NAT, not the one that could be running on the same host where FS is installed. I'll see if I can find a utility that checks that the ports are open after FS is up and running. Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26753168.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Dec 11 15:43:33 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 17:43:33 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26753168.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> <26753168.post@talk.nabble.com> Message-ID: <4C923D5E-8A20-4905-B917-F45F72300343@freeswitch.org> You don't have to do that usually... /b On Dec 11, 2009, at 5:38 PM, Fred-145 wrote: > I'll see if I can find a utility that checks that the ports are open > after > FS is up and running. From brian at freeswitch.org Fri Dec 11 15:44:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 17:44:49 -0600 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> Message-ID: %23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: > This line is basically saying that you have a call coming from > 4165551212 and it's looking for a destination number of > %23904161234. The key here is that it is coming in the public > context so you'll need to handle the routing in conf/dialplan/ > public.xml > > What should this call be doing once it comes in to FS? > > -MC From bcxml at hotmail.com Fri Dec 11 16:02:16 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 16:02:16 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> Message-ID: <26753346.post@talk.nabble.com> The version is FreeSWITCH Version 1.0.4 (14460) Brian Brian West-3 wrote: > > %23 is # so the question is should we URL decode that before routing? > I thought we did... what version are you using now? > > /b > > On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: > >> This line is basically saying that you have a call coming from >> 4165551212 and it's looking for a destination number of >> %23904161234. The key here is that it is coming in the public >> context so you'll need to handle the routing in conf/dialplan/ >> public.xml >> >> What should this call be doing once it comes in to FS? >> >> -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26753346.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Fri Dec 11 22:14:23 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 11 Dec 2009 22:14:23 -0800 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <26753346.post@talk.nabble.com> References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> <26753346.post@talk.nabble.com> Message-ID: On Dec 11, 2009, at 4:02 PM, bcxml wrote: > > The version is > > FreeSWITCH Version 1.0.4 (14460) > Ouch. You are nearly 6 months and 1500 revs behind. You badly need to update to latest trunk. -MC > >> >> >> >>> >>> >> >> From thangappan143 at gmail.com Fri Dec 11 23:38:51 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 12 Dec 2009 13:08:51 +0530 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> Message-ID: <7aa29e790912112338g23ab5867qeeca426192f34709@mail.gmail.com> Thanks for your answers. Rather than select or polling the variable on the channel, CHANNEL EXECUTE COMPLETE event is used and now it is being worked fine. ---------- Forwarded message ---------- From: Thangappan.M Date: Wed, Nov 25, 2009 at 11:48 AM Subject: Re: Problem while playing more than 10 voice files using playback To: freeswitch-users The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted > already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. > (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > >> The reason for waiting only for DTMF event is to handle the time outs in >> the IVR concept like response and inter digit time out. Using our own logic >> we 10 voice files in each play back if the voice files are more than 10. Now >> it works fine. >> >> Now the new problem has been raised. The problem is we are filtering only >> for DTMF events but we are getting COMMAND event . Because of this the DTMF >> digits are missing at the time . I am not able to proceed further. We are >> in the critical situation. >> >> Why this command event is occurring? >> How can I restrict this? >> What are the information it has? >> How can I get all the information in it ? ( If command event has info) >> >> Help me............ >> >> >> On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: >> >>> I am waiting only for DTMF events. That's why I am setting freeswitch >>> variable for knowing whether the playback has done. >>> >>> My question is "why this freeswitch variable is not setting properly when >>> I play back more than 10 files using playback_delimiter option?". >>> >>> When I play back lesser than ten voice files the variable has been set >>> properly. What could be the reason? >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Thangappan.M >>> Date: Sat, Nov 21, 2009 at 2:52 PM >>> Subject: Problem while playing more than 10 voice files using playback >>> To: freeswitch-users >>> >>> >>> Dear all, >>> >>> I am in the process of implementing IVR using event outbound >>> socket (async mode). >>> I have implemented using Perl language. >>> >>> I did the following steps: >>> => Set the playback_delimiter variable >>> => Set the playback_sleep_val variable >>> => Set the event lock as true >>> => Set the freeswitch ( my own) variable as zero >>> => Wait in the loop until the variable is been set as >>> zero >>> => Playback the voice files ( Here I combined the >>> voice files with the delimiter value if more than one voice files are there) >>> => Set the freeswitch(my own) variable as true ( This >>> is used to identify whether the voice files are played >>> successfully). >>> => Wait in the loop until the variable is been set as >>> one. >>> => Set the Event lock as false >>> >>> => Trying to get the DTMF digits ( Have a assurance >>> that all the voice files are played). >>> >>> The problem is, >>> >>> The above steps are working fine when the voice file count >>> is lesser than or equal to 10. After the voice files are played only the >>> variable(my own freeswitch) is set. Based on the variable I am doing further >>> things. >>> >>> But when I tried to give the voice files count of more than >>> 10 the variable has been set while starting to play back the first voice >>> file itself . Because of this I am not able to proceed further. >>> >>> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >>> >>> *NOTE*: I also referred mod_file_string documentation. In that they >>> specified 128 files can be used to play back the voice files using >>> playback_delimiter option. >>> >>> Please help me................? >>> Thanks in advance. >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/b2a96371/attachment-0001.html From thangappan143 at gmail.com Fri Dec 11 23:42:23 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 12 Dec 2009 13:12:23 +0530 Subject: [Freeswitch-users] Getting started on IVR Library Message-ID: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me....... -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/ae3f7a7c/attachment.html From surajee at gmail.com Fri Dec 11 23:08:13 2009 From: surajee at gmail.com (Surajee Ratnayake) Date: Sat, 12 Dec 2009 12:38:13 +0530 Subject: [Freeswitch-users] Freeswitch and Gtalk Message-ID: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Hello.. just want to get the following clarified from the friends in the same domain, since freeswitch is allowing multiple gtalk user registrations with gtalk servers, assume we route gtalk voice calls coming to these gtalk users are routed to sip extensions or to PSTN/PLMN? will google block some thing like that or is it already happening? scenario is, gtalk client A----------------------> gtalk user B at Freeswitch ------------------------------> PSTN thanx in advance, Sur -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/af3e514f/attachment.html From mike at jerris.com Sat Dec 12 00:50:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 12 Dec 2009 03:50:47 -0500 Subject: [Freeswitch-users] Getting started on IVR Library In-Reply-To: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Message-ID: A good example of how to use this code would be in mod_rss or mod_voicemail in tree. I would say look at the doxygen at http://docs.freeswitch.org/group__switch__ivr.html but it appears that page is completely broken. I will try to take a look and figure out why this weekend, in the meantime, you can look at the doxygen comments inline in switch_ivr*.h. Mike On Dec 12, 2009, at 2:42 AM, Thangappan.M wrote: > Dear all , > > I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/8f2564c9/attachment.html From mike at jerris.com Sat Dec 12 01:01:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 12 Dec 2009 04:01:51 -0500 Subject: [Freeswitch-users] Freeswitch and Gtalk In-Reply-To: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Message-ID: That should work fine. Mike On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote: > Hello.. > just want to get the following clarified from the friends in the same domain, > > since freeswitch is allowing multiple gtalk user registrations with gtalk servers, assume we route gtalk voice calls coming to these gtalk users are routed to sip extensions or to PSTN/PLMN? will google block some thing like that or is it already happening? From mailinglist at fribert.dk Sat Dec 12 03:28:37 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sat, 12 Dec 2009 12:28:37 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <117186.49301.qm@web56406.mail.re3.yahoo.com> References: <4B213D87020000E100000322@mail.fribert.dk> <117186.49301.qm@web56406.mail.re3.yahoo.com> Message-ID: <4B238C75020000E10000032D@mail.fribert.dk> Hi Mark et al. Don't know where my reply went, but certainly not to the list :-) So my public: default.xml looks like this; If I don't have any phones externally to my FS, could I just empty the public's default.xml? BR Fribert >>> 11-12-2009 kl. 10:44 skrev Mark Crane i meddelelsen <117186.49301.qm at web56406.mail.re3.yahoo.com>: What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark The extensions are located under directory/default, and they look like this: As I understand them, the context set there, is the right one? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/482f7ecf/attachment-0001.html From dftoro at yahoo.com Sat Dec 12 10:01:22 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sat, 12 Dec 2009 10:01:22 -0800 (PST) Subject: [Freeswitch-users] fifo caller id In-Reply-To: <4B238C75020000E10000032D@mail.fribert.dk> Message-ID: <319374.69139.qm@web33508.mail.mud.yahoo.com> Hello, ? I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and? fifo_consumer_caller_import variables, but i don't know use it. ? I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/b9e0539e/attachment.html From surajee at gmail.com Sat Dec 12 14:34:48 2009 From: surajee at gmail.com (Surajee Ratnayake) Date: Sun, 13 Dec 2009 04:04:48 +0530 Subject: [Freeswitch-users] Freeswitch and Gtalk In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Message-ID: <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> Thank you, I believe that will legally be okay for google also. I was worried that they might block access to our servers later Sur On 12/12/09, Michael Jerris wrote: > That should work fine. > > Mike > > On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote: > >> Hello.. >> just want to get the following clarified from the friends in the same >> domain, >> >> since freeswitch is allowing multiple gtalk user registrations with gtalk >> servers, assume we route gtalk voice calls coming to these gtalk users are >> routed to sip extensions or to PSTN/PLMN? will google block some thing >> like that or is it already happening? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From abeka at greatiam.com Sat Dec 12 15:17:46 2009 From: abeka at greatiam.com (Otis) Date: Sat, 12 Dec 2009 23:17:46 +0000 Subject: [Freeswitch-users] Link between User context and dialplan Message-ID: <4B24249A.1040004@greatiam.com> Hi folks I am so sorry if this is such a basic thing. well, when a user/extension eg 8888 is created in with say a user context - SWAHILI-SPEAKERS Please hear are my questions: 1. What dialplan will that user/extn use. 2. I guess I have to create a dialplan Should the dial-plan also be called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be any name ? 3. And how does FS know to load that dialplan for that user. 4. Where should that xml file be stored ? 5. Is there a means of determinig which dialplan was used for a call ? Thanks I think I have demonstrated enough thickness for now From dfansler at dv-fansler.com Sat Dec 12 19:28:14 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sat, 12 Dec 2009 22:28:14 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> Message-ID: <01a401ca7ba4$4e449190$eacdb4b0$@com> I am new to FreeSWITCH (ok a month old) and am still learning as hard as I can. In the recent talk about documentation, I had noticed that finding documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to download all the wiki pages and put them in some sort of logical order. No I am not finished yet, but does anyone else out there realize how many wiki pages there are on FreeSWITCH's site? So far I am up to 767 pages and have another 100 to go. I had no idea what I was getting into! David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From abeka at greatiam.com Sat Dec 12 19:40:08 2009 From: abeka at greatiam.com (Otis) Date: Sun, 13 Dec 2009 03:40:08 +0000 Subject: [Freeswitch-users] Link between Use-context and dialplan Message-ID: <4B246218.2020804@greatiam.com> Sorry I posted this earlier but did not do the due diligence and sent it with so much typo them meaning does not come out: In a nutshell I would like to know : 1. How FS would know which dialplan to use for an extension with user context other than default. 2. If a file file has to be created does the name matter 3. Where should that file be located. Thanks. From rjcajax at gmail.com Sat Dec 12 19:41:54 2009 From: rjcajax at gmail.com (Robert Clayton) Date: Sat, 12 Dec 2009 22:41:54 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <01a401ca7ba4$4e449190$eacdb4b0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: David, Thanks for your hard work. Maybe more organization will make the areas needing substance or explanation more obvious. Bob On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/01b891cb/attachment.html From msc at freeswitch.org Sat Dec 12 19:47:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Dec 2009 19:47:19 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <01a401ca7ba4$4e449190$eacdb4b0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: <343FDCB3-6D21-4D0D-B51E-8BC2C57B202F@freeswitch.org> On Dec 12, 2009, at 7:28 PM, "David V. Fansler" wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as > hard as I > can. In the recent talk about documentation, I had noticed that > finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I > decided to > download all the wiki pages and put them in some sort of logical > order. No > I am not finished yet, but does anyone else out there realize how > many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages > and have > another 100 to go. > > I had no idea what I was getting into! > Hehe neither did I when I started a few years ago. This is a monumetal task but since you've overcome the inertia let's just roll up our sleeves and get it done. I'm glad to have your help. I will contact you off list to follow up. Thanks! -MC (IRC: mercutioviz) > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Dec 12 19:48:51 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Dec 2009 19:48:51 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: On Dec 12, 2009, at 7:41 PM, Robert Clayton wrote: > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. That's my hope as well. -MC > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler > wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as > hard as I > can. In the recent talk about documentation, I had noticed that > finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I > decided to > download all the wiki pages and put them in some sort of logical > order. No > I am not finished yet, but does anyone else out there realize how > many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages > and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/e8dd5498/attachment.html From help at pdscc.com Sun Dec 13 01:53:43 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Sun, 13 Dec 2009 01:53:43 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20090823235030.A95BF5FE@sinclaire.sibble.net>, <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org> Message-ID: <20091213095350.F01371646@sinclaire.sibble.net> Brian, you have any inside info on when zfone3 is supposed to be available? Emails to the zfoneproject addresses haven't gotten me any info (or a reponse for that matter), I guess cause I am not a dev... On 23 Aug 2009 at 18:53, Brian West wrote: > Wish they would send me one for my E63 for testing... only been > working with zfone 3 so far. > > /b > > On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote: > > > Well good news for the Tiviphone client -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From dujinfang at gmail.com Sun Dec 13 04:54:53 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 13 Dec 2009 20:54:53 +0800 Subject: [Freeswitch-users] fifo caller id In-Reply-To: <319374.69139.qm@web33508.mail.mud.yahoo.com> References: <4B238C75020000E10000032D@mail.fribert.dk> <319374.69139.qm@web33508.mail.mud.yahoo.com> Message-ID: <23f91030912130454w21319313i82b21b2633cad2e5@mail.gmail.com> I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro > Hello, > > I want to know how can I get caller id after call is out queue fifo, I read > about fifo_caller_consumer_import and fifo_consumer_caller_import > variables, but i don't know use it. > > I appreciate any suggestion > > Diego Toro > http://lacarretade.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/41a3913e/attachment.html From jmesquita at freeswitch.org Sun Dec 13 06:46:36 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 13 Dec 2009 12:46:36 -0200 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <4B246218.2020804@greatiam.com> References: <4B246218.2020804@greatiam.com> Message-ID: inline.... JM On Sun, Dec 13, 2009 at 1:40 AM, Otis wrote: > Sorry > > I posted this earlier but did not do the due diligence and sent it with > so much typo them meaning does not come out: > > In a nutshell I would like to know : > > 1. How FS would know which dialplan to use for an extension with user > context other than default. > The SIP profile that the call comes in has a context. All calls that do not have users associated (not authenticated) or users that do not have the user_context var set will use that context. If the user has the user_context var set, it will use the specified one. > 2. If a file file has to be created does the name matter > No. 3. Where should that file be located. > > ${FSROOT}/conf/dialplan/* I *strongly *suggest you to read the default configs and the wiki. > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/b54a00b6/attachment.html From yehavi.bourvine at gmail.com Sun Dec 13 06:51:04 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 16:51:04 +0200 Subject: [Freeswitch-users] Sofia performance Message-ID: Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/71604c28/attachment.html From tzury.by at reguluslabs.com Sun Dec 13 06:56:26 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Sun, 13 Dec 2009 09:56:26 -0500 Subject: [Freeswitch-users] sangoma a101 - error compiling Wanpipe drivers with TDM API support on ubuntu 9.10 Message-ID: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> Hi all, I am getting a frustrating errors while trying to build the latest wanpipe on my newly and freshly installed Ubuntu 9.10 I pastebin the output at http://gist.github.com/255442 and would appreciate any help. Thanks, -- Tzury Bar Yochay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/5acb7003/attachment.html From niall.crosby at gmail.com Sun Dec 13 07:05:12 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 15:05:12 +0000 Subject: [Freeswitch-users] Java ESL Message-ID: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Hi, I am about to start writing a Java Event Socket Library as I can't find one already written thats available. 1 - Is there one already out there? 2 - If not, any pointers as to what design I should follow? Which of the current ESL's is the best modal to follow? Thanks, Niall. -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/396134a7/attachment.html From jbr at consiglia.dk Sun Dec 13 07:19:46 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 13 Dec 2009 16:19:46 +0100 Subject: [Freeswitch-users] Presence across several networked FSs Message-ID: I'm working on a setup with several networked FreeSWITCHes based on a central FS and one or more satellite FSs. The boxes are connected to each other with IAX, SIP over VPN or another protocol. In the choice of the protocol the presence issue mentioned below should be considered, and the real life practicalities such as routers, NAT and connections with packet loss. I find the directory facility as a good tool for expressing the topology on each server, where the dial-string in the directory can be used to beak out of the box into some of the other boxes via the central unit. When I dial out from a phone, the presence information is updated OK on same phone and other phones on the same box. I would like to keep track of the presence of the users on the other boxes as well. Any suggestion on to how the presence information scan be propagated to all boxes in the network. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/41ea0c39/attachment.html From frank at carmickle.com Sun Dec 13 07:21:59 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 10:21:59 -0500 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <4B246218.2020804@greatiam.com> References: <4B246218.2020804@greatiam.com> Message-ID: <20091213152158.GV31924@base.carmickle.com> On Sun, Dec 13, Otis wrote: > Sorry > > I posted this earlier but did not do the due diligence and sent it with > so much typo them meaning does not come out: > > In a nutshell I would like to know : > > 1. How FS would know which dialplan to use for an extension with user > context other than default. It just uses the context tag which you include extensions in side it. > 2. If a file file has to be created does the name matter Doesn't matter. The include statements pull the files in. Wrap the stuff in the included file in tags. > 3. Where should that file be located. Anywhere! It easiest to set up includes like Now you can put all kinds of files in the public dir and they will get included when the preprocess runs. The preprocess runs at start up so you need to restart IIRC. HTH --FC From jmesquita at freeswitch.org Sun Dec 13 07:39:06 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 13 Dec 2009 13:39:06 -0200 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Message-ID: Can't we just swig it to Java? JM On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby wrote: > > Hi, > > I am about to start writing a Java Event Socket Library as I can't find one > already written thats available. > 1 - Is there one already out there? > 2 - If not, any pointers as to what design I should follow? Which of the > current ESL's is the best modal to follow? > > Thanks, > Niall. > > -- > -- > > The information transmitted is intended only for the person or entity to > which it is addressed and may contain confidential and/or privileged > material. Statements and opinions expressed in this e-mail may not represent > those of the sender. Any review, retransmission, dissemination or other use > of, or taking of any action in reliance upon, this information by persons or > entities other than the intended recipient is prohibited. If you received > this in error, please contact the sender immediately and delete the material > from any computer. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/2628e43a/attachment-0001.html From Russell.Mosemann at cune.org Sun Dec 13 07:44:16 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Sun, 13 Dec 2009 09:44:16 -0600 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <20091213152158.GV31924@base.carmickle.com> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> Message-ID: <115561884BFA43AAA5CACA534551E5F1@cune.pri> Frank Carmickle wrote: > Now you can put all kinds of files in the public dir and they will get > included when the preprocess runs. The preprocess runs at start up so you > need to restart IIRC. Or fire up /bin/fs_cli and issue a "reloadxml" or "reload ", if a change affects a module. -- Russell Mosemann From dujinfang at gmail.com Sun Dec 13 07:47:04 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 13 Dec 2009 23:47:04 +0800 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: Message-ID: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> you can use the same ip with different port 2009/12/13, Yehavi Bourvine : > Hello, > > In the WIKI page that talks about Freeswitch performance there is a > sentence: > > *libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles* > > How can I enable more than one profile on the same interface? Won't they > colide when using the same IP and port? > > Thanks! __Yehavi: > From niall.crosby at gmail.com Sun Dec 13 07:50:28 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 15:50:28 +0000 Subject: [Freeswitch-users] Java ESL In-Reply-To: References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Message-ID: <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> Pure Java is my preference - am looking to build apps that are portable. N. 2009/12/13 Jo?o Mesquita > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby wrote: > >> >> Hi, >> >> I am about to start writing a Java Event Socket Library as I can't find >> one already written thats available. >> 1 - Is there one already out there? >> 2 - If not, any pointers as to what design I should follow? Which of the >> current ESL's is the best modal to follow? >> >> Thanks, >> Niall. >> >> -- >> -- >> >> The information transmitted is intended only for the person or entity to >> which it is addressed and may contain confidential and/or privileged >> material. Statements and opinions expressed in this e-mail may not represent >> those of the sender. Any review, retransmission, dissemination or other use >> of, or taking of any action in reliance upon, this information by persons or >> entities other than the intended recipient is prohibited. If you received >> this in error, please contact the sender immediately and delete the material >> from any computer. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/725829c6/attachment.html From yehavi.bourvine at gmail.com Sun Dec 13 08:05:17 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 18:05:17 +0200 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: I would like all phones have the same general configuration... If no other way, then I'll do that. Thanks, __Yehavi: 2009/12/13 Seven Du > you can use the same ip with different port > > 2009/12/13, Yehavi Bourvine : > > Hello, > > > > In the WIKI page that talks about Freeswitch performance there is a > > sentence: > > > > *libsofia only handles 1 thread per profile, so if that is your bottle > neck > > use more profiles* > > > > How can I enable more than one profile on the same interface? Won't they > > colide when using the same IP and port? > > > > Thanks! __Yehavi: > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/9c6592df/attachment.html From moises.silva at gmail.com Sun Dec 13 08:19:56 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 13 Dec 2009 11:19:56 -0500 Subject: [Freeswitch-users] sangoma a101 - error compiling Wanpipe drivers with TDM API support on ubuntu 9.10 In-Reply-To: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> References: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> Message-ID: On Sun, Dec 13, 2009 at 9:56 AM, Tzury Bar Yochay wrote: > Hi all, > > I am getting a frustrating errors while trying to build the latest wanpipe > on my newly and freshly installed Ubuntu 9.10 > > I pastebin the output at http://gist.github.com/255442 and would > appreciate any help. > > Hi Tzury, It seems the kernel developers decided to move some members of the net_device kernel structure to an internal structure called net_device_ops netdev_ops, I see Ubuntu 9.10 uses a pretty recent kernel. The engineer in charge of the wanpipe drivers will likely soon put a fix. In the meantime you will have to downgrade the kernel somehow. It is usually recommended to use CentOS for servers running wanpipe, I strongly recommend you to switch if possible. If changing to CentOS is not an option, you will have to wait for the wanpipe drivers to be fixed for this recent kernel. You can always send an e-mail to support at sangoma.com to let them know and that may speed things up. PD. I will answer your other e-mail soon. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/8cff8a7c/attachment.html From abeka at greatiam.com Sun Dec 13 08:19:05 2009 From: abeka at greatiam.com (Otis) Date: Sun, 13 Dec 2009 16:19:05 +0000 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <20091213152158.GV31924@base.carmickle.com> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> Message-ID: <4B2513F9.4020000@greatiam.com> Frank Carmickle wrote: > On Sun, Dec 13, Otis wrote: > >> Sorry >> >> I posted this earlier but did not do the due diligence and sent it with >> so much typo them meaning does not come out: >> >> In a nutshell I would like to know : >> >> 1. How FS would know which dialplan to use for an extension with user >> context other than default. >> > > It just uses the context tag which you include extensions in side it. > > > > > > > > > > >> 2. If a file file has to be created does the name matter >> > > Doesn't matter. The include statements pull the files in. Wrap the stuff in the included file in tags. > > >> 3. Where should that file be located. >> > > Anywhere! It easiest to set up includes like > > > > Now you can put all kinds of files in the public dir and they will get included when the preprocess runs. The preprocess runs at start up so you need to restart IIRC. > > HTH > --FC > > > Thank you so much for your time. From mrene_lists at avgs.ca Sun Dec 13 08:26:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 13 Dec 2009 11:26:11 -0500 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: <57B69B54-9477-4AEB-B16F-DA235C646A48@avgs.ca> You can have multiple SRV records pointing to different ports. See: http://mit.edu/sip/sip.edu/dns.shtml Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Dec-09, at 11:05 AM, Yehavi Bourvine wrote: > I would like all phones have the same general configuration... If no > other way, then I'll do that. > > Thanks, __Yehavi: > > 2009/12/13 Seven Du > you can use the same ip with different port > > 2009/12/13, Yehavi Bourvine : > > Hello, > > > > In the WIKI page that talks about Freeswitch performance there > is a > > sentence: > > > > *libsofia only handles 1 thread per profile, so if that is your > bottle neck > > use more profiles* > > > > How can I enable more than one profile on the same interface? > Won't they > > colide when using the same IP and port? > > > > Thanks! __Yehavi: > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/d1db0855/attachment-0001.html From frank at carmickle.com Sun Dec 13 08:29:32 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 11:29:32 -0500 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: <20091213162932.GW31924@base.carmickle.com> On Sun, Dec 13, Yehavi Bourvine wrote: > I would like all phones have the same general configuration... If no other > way, then I'll do that. Have you already set up a system and found the load of all your phones to be to high? How many phones are we talking about? A load balancer is a solution if you've already tweaked the system for maximum performance. --FC From frank at carmickle.com Sun Dec 13 08:35:56 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 11:35:56 -0500 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <115561884BFA43AAA5CACA534551E5F1@cune.pri> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> <115561884BFA43AAA5CACA534551E5F1@cune.pri> Message-ID: <20091213163556.GX31924@base.carmickle.com> On Sun, Dec 13, Russell Mosemann wrote: > Frank Carmickle wrote: > > Now you can put all kinds of files in the public dir and they will get > > included when the preprocess runs. The preprocess runs at start up so you > > need to restart IIRC. > > Or fire up /bin/fs_cli and issue a "reloadxml" or "reload ", if a change affects a module. Thanks for clearing that up for me. I seemed to remember that the wiki was unclear about preprocessor variables being updated with a reloadxml but it is clear now if it wasn't before. --FC From yehavi.bourvine at gmail.com Sun Dec 13 10:21:45 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 20:21:45 +0200 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <20091213162932.GW31924@base.carmickle.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> Message-ID: We are still on a small "proof of concept" system, but I am looking at the future... Thanks, __Yehavi: 2009/12/13 Frank Carmickle > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all phones have the same general configuration... If no > other > > way, then I'll do that. > > Have you already set up a system and found the load of all your phones to > be to high? How many phones are we talking about? A load balancer is a > solution if you've already tweaked the system for maximum performance. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/3716d911/attachment.html From anthony.minessale at gmail.com Sun Dec 13 10:29:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 12:29:20 -0600 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> Message-ID: <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> The swig java one is almost done we need someone who likes java to finish it but as you can see most java ppl seem to always want to do it "their way" On Dec 13, 2009 9:56 AM, "Niall Crosby" wrote: Pure Java is my preference - am looking to build apps that are portable. N. 2009/12/13 Jo?o Mesquita > > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> Message-ID: <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> Here is my standard asvice on people worrying about performance before even trying fs. Rule of thumb, if you have ever used asterisk, multiply everything by 10 so if there is a performance concern assume it will not arise unless you get at least 10 asterisks worth of performance first. People do thosands of channels with media and tens of thousands with no media, try it first before freting about imaginary load concerns. On Dec 13, 2009 12:28 PM, "Yehavi Bourvine" wrote: We are still on a small "proof of concept" system, but I am looking at the future... Thanks, __Yehavi: 2009/12/13 Frank Carmickle > > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all phones have the same general config... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/5345bf5f/attachment.html From niall.crosby at gmail.com Sun Dec 13 10:42:56 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 18:42:56 +0000 Subject: [Freeswitch-users] Java ESL In-Reply-To: <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> Message-ID: <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> Where can I find the Java swig ESL? I like Java so am happy to put time towards generating a pure Java ECL, however I haven't programmed C in 10+ years so feel like swig would be to much in the deep end for me. 2009/12/13 Anthony Minessale > The swig java one is almost done we need someone who likes java to finish > it but as you can see most java ppl seem to always want to do it "their way" > > On Dec 13, 2009 9:56 AM, "Niall Crosby" wrote: > > > Pure Java is my preference - am looking to build apps that are portable. > > N. > > 2009/12/13 Jo?o Mesquita > > > > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 > PM, Niall Crosby > -- -- The information transmitted is intended only for the person or entity > to which it is add... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/72d39367/attachment.html From jaybinks at gmail.com Sun Dec 13 13:57:39 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 14 Dec 2009 07:57:39 +1000 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> Message-ID: <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> I'm interested in what the upper limit would be, when expecting a performance improvement with sofia profiles. For example let's say I were to direct connect to customers ( layer 2 ) with a .1q trunk coming in to fs and a Sofia profile for each customer. Am I going to hit a bottleneck at 20,50,100,500 ??? Guess it's hardware limited , but any thoughts ? J On 14/12/2009, at 4:36, Anthony Minessale wrote: > Here is my standard asvice on people worrying about performance > before even trying fs. > > Rule of thumb, if you have ever used asterisk, multiply everything > by 10 so if there is a performance concern assume it will not arise > unless you get at least 10 asterisks worth of performance first. > > People do thosands of channels with media and tens of thousands with > no media, try it first before freting about imaginary load concerns. > >> On Dec 13, 2009 12:28 PM, "Yehavi Bourvine" > > wrote: >> >> We are still on a small "proof of concept" system, but I am >> looking at the future... >> >> Thanks, __Yehavi: >> >> 2009/12/13 Frank Carmickle >> > > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all >> phones have the same general config... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/1cc75737/attachment-0001.html From quentusrex at gmail.com Sun Dec 13 14:33:09 2009 From: quentusrex at gmail.com (William King) Date: Sun, 13 Dec 2009 14:33:09 -0800 Subject: [Freeswitch-users] CDR question. Message-ID: <4B256BA5.2050006@gmail.com> Anyone know a good way to determine which extension picked up a call that was bridged to 10+ extensions? -William From anthony.minessale at gmail.com Sun Dec 13 14:37:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 16:37:24 -0600 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> Message-ID: <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> Sep processes does better than sep profiles. We need to push the sofia devs to work on a better concurrancy scheme but they are too busy with other nokia duties these days so were stuck with what we got for now. About 400cps on a good day On Dec 13, 2009 4:05 PM, "Jay Binks" wrote: I'm interested in what the upper limit would be, when expecting a performance improvement with sofia profiles. For example let's say I were to direct connect to customers ( layer 2 ) with a .1q trunk coming in to fs and a Sofia profile for each customer. Am I going to hit a bottleneck at 20,50,100,500 ??? Guess it's hardware limited , but any thoughts ? J On 14/12/2009, at 4:36, Anthony Minessale wrote: > Here is my standa... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/2eb5b1b8/attachment.html From anthony.minessale at gmail.com Sun Dec 13 14:38:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 16:38:41 -0600 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> Message-ID: <191c3a030912131438u56863e73h3f95f38b0e536848@mail.gmail.com> In the libs/esl there is already swigged java but I don't know how to load it etc try make javamod in esl On Dec 13, 2009 12:47 PM, "Niall Crosby" wrote: Where can I find the Java swig ESL? I like Java so am happy to put time towards generating a pure Java ECL, however I haven't programmed C in 10+ years so feel like swig would be to much in the deep end for me. 2009/12/13 Anthony Minessale > > The swig java one is almost done we need someone who likes java to finish it but as you can see... -- -- The information transmitted is intended only for the person or entity to which it is addressed and ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/74566495/attachment.html From dfansler at dv-fansler.com Sun Dec 13 17:27:39 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sun, 13 Dec 2009 20:27:39 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: <02cd01ca7c5c$a05b64a0$e1122de0$@com> Final Count was just over 900 files. At the moment I am putting them in a logical order - as best I can tell with my limited experience - combining chapters and providing links from the table of contents to sections in each chapter. Then I will go back in retarget all the hyperlinks to point to the document rather than the wiki site. This may take a few hours J David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert Clayton Sent: Saturday, December 12, 2009 10:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch Documentation David, Thanks for your hard work. Maybe more organization will make the areas needing substance or explanation more obvious. Bob On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler wrote: I am new to FreeSWITCH (ok a month old) and am still learning as hard as I can. In the recent talk about documentation, I had noticed that finding documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to download all the wiki pages and put them in some sort of logical order. No I am not finished yet, but does anyone else out there realize how many wiki pages there are on FreeSWITCH's site? So far I am up to 767 pages and have another 100 to go. I had no idea what I was getting into! David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/f708e9cc/attachment.html From jingwei.yang at gmail.com Sun Dec 13 18:11:38 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 14 Dec 2009 10:11:38 +0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: <13529f9d0912131811r5ea76df9g37d961ff949dd5e5@mail.gmail.com> Thanks David. It'll be of a great help to FS newbies like me. Regards, -Jingwei On Mon, Dec 14, 2009 at 9:27 AM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Robert > Clayton > *Sent:* Saturday, December 12, 2009 10:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch Documentation > > > > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler < > dfansler at dv-fansler.com> wrote: > > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/89dcd08f/attachment-0001.html From dome at tel.co.th Sun Dec 13 18:41:57 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 14 Dec 2009 09:41:57 +0700 Subject: [Freeswitch-users] Event Socket outbound in PHP Message-ID: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> Dear All, Now i use php for ESL outbound. i get variable from stdin and process. (i use xinetd for handle socket) $in = fopen("php://stdin", "r"); Problem is when i use read command for get input from DTMF. i can't get variable. So now i use 2 php script. and use read appliction in XML DIalplan for solve this problem. I plan to use php handle socket like a perl in http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl But i want to know how PHP work like this example ? my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); Can someoue help me ? Best Regards. Dome C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b5d489b5/attachment.html From thangappan143 at gmail.com Sun Dec 13 21:10:28 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 14 Dec 2009 10:40:28 +0530 Subject: [Freeswitch-users] Fwd: Getting started on IVR Library In-Reply-To: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Message-ID: <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> I've seen the source code of mod_rss and mod_voicemail. I was not able to get it. What are the steps do I need to follow for implementing IVR using IVR library? Is there any documentations available for knowing about IVR library? I might be wrong please correct me... Please help me............ ---------- Forwarded message ---------- From: Thangappan.M Date: Sat, Dec 12, 2009 at 1:12 PM Subject: Getting started on IVR Library To: freeswitch-users Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me....... -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0a841290/attachment.html From info at daccii.it Mon Dec 14 02:39:50 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 14 Dec 2009 11:39:50 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211024228.GE14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> <20091211024228.GE14547@sys11.in.vpac.org> Message-ID: <4B2615F6.7080701@daccii.it> Hi, you could build an "embedded filesystem" using one of these great tools: - buildroot crosstools-ng - ptxdist - moblin2 - clfs The one i prefer is buildroot because is simply, flexible and fast to build a root filesystem: to setup a minimal system, once you builded the toolchain using crosstools-ng, you need only a couple of hours! It support a lot of hardware platform and br mailing list is VERY active! If you need to use the hardware supplier toolchain you can, just set it instead to build one with crosstools-ng! Freeswitch uses threads a lot, more than asterisk i think, so is preferable to use glibc or eglibc (more configurable than glibc, but essentially the same) instead of uclibc because these libraries are great for embedded hardware but support only old style threads (linuxthread instead of nptl). You can use busybox as base for a lot of stuff and then build what you need (php with fastcgi, lighttpd [althought i prefer nginx], sqlite, lua and so on). I'm doing some job integrating fs with an own software, so i'll start to work on it, probably, the next year, at the end of first quarter or at start of the second, until that moment i'll use a shrinked down ubuntu 8.04 to fit it on a 4gb cf. (i got my alix board a couple of days ago!). However i've used br in a couple of personal project and to help my brother at university (needed to setup a system on a xilinx virtex ii board) If you prefer something more well tested you can give a try to ptxdist or moblin2 but them are harder to use and configure althought them works better! Best Regards, Daniele Brian May ha scritto: > On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote: >> As a note, we are pretty aggressive about making sure all this stuff works >> right out of svn without any patches so it should be easy to port freeswitch >> to most platforms now. > > Thats good to hear. > > I am guessing this means I should use a recent version. I see there is an > Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should > be able to build from the source. > > Anyway I sent an email to Yawarra to ask them if the net5501 computer > is compatible with > the TDM400 cards. There is something about a kit for the dual rack mount > computer for the TDM400, which would be good if I had a rack, and somewhere to > put a rack. So presumably this means it should work for the non-rack mount > system too. -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/bb88053f/attachment.vcf From Prometheus001 at gmx.net Mon Dec 14 04:05:32 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 14 Dec 2009 13:05:32 +0100 Subject: [Freeswitch-users] Which ATAs to chose for modem connections? Message-ID: <4B262A0C.4070604@gmx.net> We currently use Patton gateways SN4116 for attaching fax and modem equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, so media flow goes the following way: Modem/Fax => Patton_SN4116 => Patton_SN46XX =>PSTN/ISDN However modem connections are not very reliable. We exchanged the SN4118 against a "Fritzbox" ATA and the situation improves. However Fritzboxes do not deliver the number of ports we need. What is your experience? Which ATA is the best choice for modem connections? Best regards Peter From steveu at coppice.org Mon Dec 14 04:29:02 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 14 Dec 2009 20:29:02 +0800 Subject: [Freeswitch-users] Which ATAs to chose for modem connections? In-Reply-To: <4B262A0C.4070604@gmx.net> References: <4B262A0C.4070604@gmx.net> Message-ID: <4B262F8E.7040003@coppice.org> On 12/14/2009 08:05 PM, Peter P GMX wrote: > We currently use Patton gateways SN4116 for attaching fax and modem > equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, > so media flow goes the following way: > Modem/Fax => Patton_SN4116 => Patton_SN46XX =>PSTN/ISDN > However modem connections are not very reliable. We exchanged the SN4118 > against a "Fritzbox" ATA and the situation improves. However Fritzboxes > do not deliver the number of ports we need. > > What is your experience? Which ATA is the best choice for modem connections? > Best regards > Peter > > Your improvement is probably more due to luck than engineering. Turning off any dynamic jitter buffering, and making sure you use only A-law or u-law, will reduce the failures. the only thing that will give reliable results for modems is a managed, absolutely lossless, packet path, and terminal equipment which does doesn't slip data samples to allow for mismatched sampling clocks. Good luck finding those. Steve From oscav at hotmail.fr Mon Dec 14 04:45:19 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 14 Dec 2009 04:45:19 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? Message-ID: <26777181.post@talk.nabble.com> Hi, What are the solutions to support the G729/G723 codec within FreeSwitch ? Thanks -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Mon Dec 14 05:53:57 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 14 Dec 2009 05:53:57 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? Message-ID: <26778101.post@talk.nabble.com> Hello I'm a bit confused at the difference between those two concepts. Contexts are created in the /dialplan, and are refered to by items in /SIP_profiles and extensions in /directory. What purpose do contexts and profiles play? Thank you. -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26778101.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Dec 14 07:19:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:19:56 -0600 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26777181.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> Message-ID: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Software G729 will be available by the end of the month. As for, G723 we are not currently working on it. On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: > > Hi, > > What are the solutions to support the G729/G723 codec within FreeSwitch ? > > Thanks > > > -- > View this message in context: > http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/d28816c5/attachment.html From anthony.minessale at gmail.com Mon Dec 14 07:29:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:29:20 -0600 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <26778101.post@talk.nabble.com> References: <26778101.post@talk.nabble.com> Message-ID: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Profile is a collection of preferences uses by conferences etc. In the case of SIP a profile is also the name for the resulting SIP UA created by a particular profile. Context is a narrowed down view of something, in the case of the dialplan a context is a set of extensions. It's like having a dedicated set of extensions per distinct context name like parallel universes. both the foo context and the bar context can have extension 2001. On Mon, Dec 14, 2009 at 7:53 AM, Fred-145 wrote: > > Hello > > I'm a bit confused at the difference between those two concepts. Contexts > are created in the /dialplan, and are refered to by items in /SIP_profiles > and extensions in /directory. > > What purpose do contexts and profiles play? > > Thank you. > -- > View this message in context: > http://old.nabble.com/Context-vs.-profile--tp26778101p26778101.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b3d17059/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 14 07:37:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:37:32 -0600 Subject: [Freeswitch-users] Link between User context and dialplan In-Reply-To: <4B24249A.1040004@greatiam.com> References: <4B24249A.1040004@greatiam.com> Message-ID: <191c3a030912140737ldb2dbe7of181c4b6cfb22f79@mail.gmail.com> a dial plan is a another level of indirection on top of contexts it denotes a specific module which implements the entire universe of dialing with room for as many contexts as you have room for. There is an XML dialplan, an ENUM dialplan etc. You can write your own dialplan and send calls to it. On Sat, Dec 12, 2009 at 5:17 PM, Otis wrote: > Hi folks > > I am so sorry if this is such a basic thing. > > well, when a user/extension eg 8888 is created in with say a user > context - SWAHILI-SPEAKERS Please hear are my questions: > > 1. What dialplan will that user/extn use. > 2. I guess I have to create a dialplan Should the dial-plan also be > called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be > any name ? > 3. And how does FS know to load that dialplan for that user. > 4. Where should that xml file be stored ? > 5. Is there a means of determinig which dialplan was used for a call ? > > Thanks I think I have demonstrated enough thickness for now > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/92a80dfa/attachment.html From helmut.kuper at ewetel.de Mon Dec 14 08:00:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Dec 2009 17:00:03 +0100 Subject: [Freeswitch-users] bidirectional SRTP Message-ID: <4B266103.7060305@ewetel.de> Hello, today I found that my SRTP calls are only SRTP in one way (from snom to FS). The other direction is still RTP (at least if I believe wireshark). How can I get both directions using SRTP? I do a simple call to an announcement in FS. My SIP-profile has: I followed the document of FS Wiki: http://wiki.freeswitch.org/wiki/Secure_RTP But it seems there is no ch_var like sip_has_crypto. Is there a way to have both directions/streams per leg using SRTP? Please find attached the pcap trace of my snom phone. There you can see that snom offers crypto to FS. FS sends an OK with SRTP and RTP. When you look into the RTP data you see SRTP from snom to FS and RTP from FS to snom. Any hints? regards Helmut -------------- next part -------------- A non-text attachment was scrubbed... Name: trace.pcap Type: application/octet-stream Size: 79985 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/323eb17d/attachment-0001.obj From steveu at coppice.org Mon Dec 14 08:02:40 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 15 Dec 2009 00:02:40 +0800 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Message-ID: <4B2661A0.6050208@coppice.org> On 12/14/2009 11:19 PM, Anthony Minessale wrote: > Software G729 will be available by the end of the month. > As for, G723 we are not currently working on it. There is a legit option for G.723.1 - the Digium TC400B card. Its supported by Freeswitch, thanks to Moises. > > > On Mon, Dec 14, 2009 at 6:45 AM, Oscav > wrote: > > > Hi, > > What are the solutions to support the G729/G723 codec within > FreeSwitch ? > Steve From mrene_lists at avgs.ca Mon Dec 14 08:05:47 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 14 Dec 2009 11:05:47 -0500 Subject: [Freeswitch-users] bidirectional SRTP In-Reply-To: <4B266103.7060305@ewetel.de> References: <4B266103.7060305@ewetel.de> Message-ID: <36F785FB-8FCF-436C-BB80-9C9B34F013CF@avgs.ca> Both the INVITE and the 200 OK have an a=crypto line. You need to know that only the rtp payload is encrypted, it is normal that you see the headers as-is. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 14-Dec-09, at 11:00 AM, Helmut Kuper wrote: > Hello, > > > today I found that my SRTP calls are only SRTP in one way (from snom > to FS). The other direction is still RTP (at least if I believe > wireshark). How can I get both directions using SRTP? > > I do a simple call to an announcement in FS. > > My SIP-profile has: > > > I followed the document of FS Wiki: > http://wiki.freeswitch.org/wiki/Secure_RTP > > But it seems there is no ch_var like sip_has_crypto. > > Is there a way to have both directions/streams per leg using SRTP? > > Please find attached the pcap trace of my snom phone. > > There you can see that snom offers crypto to FS. FS sends an OK with > SRTP and RTP. When you look into the RTP data you see SRTP from snom > to FS and RTP from FS to snom. > > > Any hints? > > regards > Helmut > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Mon Dec 14 09:12:48 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 14 Dec 2009 09:12:48 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Message-ID: <26779926.post@talk.nabble.com> Thanks Anthony for the tip. Would you say this is a correct representation of things? "Contexts are a set of extensions in conf/dialplan/ (eg. default, public, etc.) Extensions are configured through files in conf/directory/. Each extension maps to a context (). Profiles refer to contexts in a dialplan (eg. ). Profiles are groups of settings used by different parts of the network, eg. Internal (private LAN), External (Internet-accessible, public LAN), etc. Each profile has a unique IP + port number. " -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26779926.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Dec 14 09:21:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 09:21:05 -0800 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Message-ID: <87f2f3b90912140921t76149076y505e4dfc5eefdb76@mail.gmail.com> Profile is a collection of preferences uses by conferences etc. In the case of SIP a profile is also the name for the resulting SIP UA created by a particular profile. Context is a narrowed down view of something, in the case of the dialplan a context is a set of extensions. It's like having a dedicated set of extensions per distinct context name like parallel universes. both the foo context and the bar context can have extension 2001. A classic example of this is with the default dialing range of 1000-1019. This range appears in all three contexts that are defined in the default configuration: conf/dialplan/public.xml in "public_extensions" conf/dialplan/features.xml in "please_hold" conf/dialplan/default.xml in "Local_Extension" If a call comes in on the public context and gets routed to 1000 (or 1001, 1002,...1019) then it gets handled by "public_extensions" in public.xml. That extension transfers the call to 1000 in the features context (defined in features.xml) which executes a "please hold while I transfer your call" kind of operation and then transfers the call to 1000 in the default context (which is defined in default.xml). Check out those three files. You'll see some cool stuff! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0de43ca5/attachment.html From msc at freeswitch.org Mon Dec 14 09:57:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 09:57:58 -0800 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <26779926.post@talk.nabble.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> <26779926.post@talk.nabble.com> Message-ID: <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> On Mon, Dec 14, 2009 at 9:12 AM, Fred-145 wrote: > > Thanks Anthony for the tip. > > Would you say this is a correct representation of things? > > "Contexts are a set of extensions in conf/dialplan/ (eg. default, public, > etc.) > > Extensions are configured through files in conf/directory/. Each extension > maps to a context (). > It would be more correct to say that "users are configured in conf/directory" > > Profiles refer to contexts in a dialplan (eg. ). > Profiles are groups of settings used by different parts of the network, eg. > A better way to say this would be that a "SIP profile defines a SIP *User Agent*." (If you don't know what a SIP user agent is that's okay - just know that a profile listens for connections on a particular IP + Port and also sends outbound traffic via the same IP/port.) A SIP profile routes unauth'd incoming calls to a pre-defined dialplan and context. Users who have auth'd to the SIP profile have a little more control - the user_context can be defined at the user level in the directory. (Normally a user will just use the "default" context but it needn't be that way - you could have a context for two different entities, e.g. two different businesses running on the same FS server.) Are you typing up something for posterity's sake? If so let me know. I'll be happy to proof-read the finished product and offer suggestions. -MC Internal (private LAN), External (Internet-accessible, public LAN), etc. > Each profile has a unique IP + port number. " > -- > View this message in context: > http://old.nabble.com/Context-vs.-profile--tp26778101p26779926.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/c7a3fff1/attachment.html From ken at ksac.com Mon Dec 14 10:45:51 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 14 Dec 2009 10:45:51 -0800 Subject: [Freeswitch-users] monday build Message-ID: HI I tried to build the svn last Friday and it didn't make the sphinx dll, so I thought I would wait for the Monday build (web site says update every Monday). Is there going to be a windows build today, and or is the sphinx dll build problem fixed if I build it myself? Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/3d4fa9dd/attachment.html From brian at freeswitch.org Mon Dec 14 10:50:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 12:50:52 -0600 Subject: [Freeswitch-users] monday build In-Reply-To: References: Message-ID: <727A1FFD-7408-4A4A-8A65-775D3DC34E7C@freeswitch.org> I do pre releases and it'll be up shortly had to fix a couple of bugs. I don't do binary releases for windows you'll have to do that yourself or wait. /b On Dec 14, 2009, at 12:45 PM, Kendall Stauffer wrote: > HI > I tried to build the svn last Friday and it didn?t make the > sphinx dll, so I thought I would wait for the Monday build (web site > says update every Monday). > Is there going to be a windows build today, and or is the > sphinx dll build problem fixed if I build it myself? > > Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/48946613/attachment-0001.html From lon at kickasspixels.com Mon Dec 14 11:11:39 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 14 Dec 2009 11:11:39 -0800 Subject: [Freeswitch-users] Billing solutions information Message-ID: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> Hey everyone, I am researching billing solutions for Freeswitch and want to consolidate the information with what others have found, then add it to the Wiki. There are seems to be a number of billing solutions by commercial providers, claiming they can integrate with Freeswitch, but nothing concrete explaining how far they go. Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, etc? Mod_nibblebill handles the basics of updating a database table. The A2Billing teams says they are planning on adding support for Freeswitch in a few months. ASTPP.org says they support Freeswitch, but the site hasn't been updated since 2008. If you know about any solutions, links to solutions or any information can you send it to me? I will organize it and add it to the wiki. Thanks! Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/604b4b2c/attachment.html From dftoro at yahoo.com Mon Dec 14 11:13:17 2009 From: dftoro at yahoo.com (Diego Toro) Date: Mon, 14 Dec 2009 11:13:17 -0800 (PST) Subject: [Freeswitch-users] fifo caller id In-Reply-To: <23f91030912130454w21319313i82b21b2633cad2e5@mail.gmail.com> Message-ID: <27104.38859.qm@web33501.mail.mud.yahoo.com> Thanks for your answer, I want to resolver the caller id to dialplan level, I know variables fifo_caller_consumer_import and fifo_consumer_caller_import but isn't clear for me how use it. Diego Toro http://lacarretade.blogspot.com/ --- On Sun, 12/13/09, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] fifo caller id To: freeswitch-users at lists.freeswitch.org Date: Sunday, December 13, 2009, 7:54 AM I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro Hello, ? I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and? fifo_consumer_caller_import variables, but i don't know use it. ? I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0774cb53/attachment.html From jeff at jefflenk.com Mon Dec 14 11:25:03 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 14 Dec 2009 11:25:03 -0800 (PST) Subject: [Freeswitch-users] monday build In-Reply-To: References: Message-ID: <1260818703016-4166167.post@n2.nabble.com> Please post back to the list if you have problems with the windows build! Everything is working as far as I know. If you have an existing build you should delete the following directories and let the scripts download it again. libs\pocketsphinx-0.5.99 <- delete libs\sphinxbase-0.4.99 <- delete Kendall Stauffer wrote: > > HI > I tried to build the svn last Friday and it didn't make the sphinx dll, > so I thought I would wait for the Monday build (web site says update every > Monday). > Is there going to be a windows build today, and or is the sphinx dll > build problem fixed if I build it myself? > > Thanks!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/monday-build-tp4166045p4166167.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Dec 14 11:45:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 11:45:12 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5pre9 is now available Message-ID: <87f2f3b90912141145t6c193b52yf464f208db7b5d59@mail.gmail.com> FYI, The latest pre-release is now available. Usual information is available here: http://www.freeswitch.org/node/222 Please update as soon as you can. (SVN trunk users do the "make current" thing please.) We need your testing and feedback please! Many thanks for continuing to support FreeSWITCH. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/34da2dd4/attachment.html From dmitry.bely at gmail.com Mon Dec 14 11:47:08 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Mon, 14 Dec 2009 22:47:08 +0300 Subject: [Freeswitch-users] Language settings for demo IVR Message-ID: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> I'm playing with demo IVR from FreeSwitch distribution and have a problem with language settings. I would like to use Russian as a default language for voice messages so I set in vars.xml and installed Russian sound files. It works almost correctly: all phrases are played in Russian, but not explicitly specified .wav files; say for Hello everyone, I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like to connect to the event socket and run some api commands and return them (as opposed to checking SIP, for example). I haven't found anything and I've started to write one in perl using ESL. I'm sure whatever I come up with is going to be pretty ugly and I'd much rather use something else. Has this been done? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Dec 14 11:50:45 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 13:50:45 -0600 Subject: [Freeswitch-users] monday build In-Reply-To: <1260818703016-4166167.post@n2.nabble.com> References: <1260818703016-4166167.post@n2.nabble.com> Message-ID: Also Pre9 is up now. /b On Dec 14, 2009, at 1:25 PM, Jeff Lenk wrote: > > Please post back to the list if you have problems with the windows > build! > Everything is working as far as I know. > If you have an existing build you should delete the following > directories > and let the scripts download it again. > > libs\pocketsphinx-0.5.99 <- delete > libs\sphinxbase-0.4.99 <- delete From msc at freeswitch.org Mon Dec 14 12:02:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:02:58 -0800 Subject: [Freeswitch-users] FreeSWITCH Nagios Plugin In-Reply-To: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com> References: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com> Message-ID: <87f2f3b90912141202n20c54caet6d0eb8d7b0af7601@mail.gmail.com> On Mon, Dec 14, 2009 at 11:49 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like > to connect to the event socket and run some api commands and return > them (as opposed to checking SIP, for example). I haven't found > anything and I've started to write one in perl using ESL. I'm sure > whatever I come up with is going to be pretty ugly and I'd much rather > use something else. Has this been done? > > Thanks! > Kristian, I started something similar a few months back before getting side-tracked. It's the skeleton for a nagios plugin, also written in Perl. Email me off list if you want to have a look... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b88d918b/attachment.html From msc at freeswitch.org Mon Dec 14 12:20:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:20:10 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: <87f2f3b90912141220w12083098v8b71559c0640a936@mail.gmail.com> On Sun, Dec 13, 2009 at 5:27 PM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > Thanks for this hard work. Please contact me if/when you need another pair of eyes on this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0389c3be/attachment-0001.html From msc at freeswitch.org Mon Dec 14 12:29:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:29:27 -0800 Subject: [Freeswitch-users] Fwd: Getting started on IVR Library In-Reply-To: <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> Message-ID: <87f2f3b90912141229n3c8edb18r4f3054732759e988@mail.gmail.com> On Sun, Dec 13, 2009 at 9:10 PM, Thangappan.M wrote: > > I've seen the source code of mod_rss and mod_voicemail. I was not able to > get it. > > What are the steps do I need to follow for implementing IVR using IVR > library? > Is there any documentations available for knowing about IVR library? > > I might be wrong please correct me... > > Please help me............ > > What kind of application are you building that requires the use of C? The reason I ask is that the higher level scripting languages and the XML configs allow you to create IVRs with much less coding. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/063d90f0/attachment.html From msc at freeswitch.org Mon Dec 14 12:58:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:58:55 -0800 Subject: [Freeswitch-users] IVR apps in lua In-Reply-To: References: Message-ID: <87f2f3b90912141258v7c66abaald4282fd33d7b7705@mail.gmail.com> On Fri, Dec 4, 2009 at 7:58 AM, Neil Patel wrote: > Hi All, > > I haven't found a substantial example of IVR applications implemented in > lua. Can anyone suggest where to look? My issue has to do with appropriate > coding style. > > I am implementing a voice message board application in lua. I want to allow > the user to dial buttons to navigate forward and back in the list of > messages. One way to implement playmessage() is to check for a forward/back > command while playing the current message, and if a command is given to > invoke playmessage() with the prev/next message in the list. However, this > leaves a chain of unreturned playmessage calls on the execution stack (a > recursive function). > > Alternatively, the playmessage() function can return control to its caller > (perhaps a while loop that spins forever) and pass back a code to indicate > the command. The caller acts accordingly. This is non-recursive, but for > anything but simple applications this style becomes tedious as you start > needing to pass back more info and up longer chains of functions. > > Any guidance on this would be appreciated. > > Thanks, > Neil > Sorry for the late response. Tony and I have just gotten to the point in the book we're writing that deals with IVRs so I've started looking at this much more closely. Lua (and Perl, PHP, and the other swig'd langs) support a light OOP way of defining IVR menus (via the IVRMenu class, btw) in a dynamic way. I've got the demo IVR all converted to a Lua script with the exception that I've got a bug to track down when using "menu-sub" as an action. (Says that the menu is invalid when I try it; I am gonna work with bkw when we get a few minutes...) Anyway, here's what I've got so far and I intend to drop it into the scripts/lua directory in the source tree once we confirm that it all works. -- lua_ivr.lua -- -- This script is virtually identical to the demo_ivr defined in conf/autoload_configs/ivr.conf.xml -- It uses the same sound files and mostly the same settings -- It is intended to be used as an example of how you can use Lua to create dynamic IVRs -- -- This hash defines the main IVR menu. It is equivalent to the lines in ivr.conf.xml ivr_def = { ["main"] = undef, ["name"] = "demo_ivr_lua", ["greet_long"] = "phrase:demo_ivr_main_menu", ["greet_short"] = "phrase:demo_ivr_main_menu_short", ["invalid_sound"] = "ivr/ivr-that_was_an_invalid_entry.wav", ["exit_sound"] = "voicemail/vm-goodbye.wav", ["confirm_macro"] = "", ["confirm_key"] = "", ["tts_engine"] = "flite", ["tts_voice"] = "rms", ["confirm_attempts"] = "3", ["inter_digit_timeout"] = "2000", ["digit_len"] = "4", ["timeout"] = "10000", ["max_failures"] = "3", ["max_timeouts"] = "2" } -- top is an object of class IVRMenu -- pass in all 16 args to the constructor to define a new IVRMenu object top = freeswitch.IVRMenu( ivr_def["main"], ivr_def["name"], ivr_def["greet_long"], ivr_def["greet_short"], ivr_def["invalid_sound"], ivr_def["exit_sound"], ivr_def["confirm_macro"], ivr_def["confirm_key"], ivr_def["tts_engine"], ivr_def["tts_voice"], ivr_def["confirm_attempts"], ivr_def["inter_digit_timeout"], ivr_def["digit_len"], ivr_def["timeout"], ivr_def["max_failures"], ivr_def["max_timeouts"] ); -- bindAction args = action, param, digits -- The following bindAction line is the equivalent of this XML from demo_ivr in ivr.conf.xml -- top:bindAction("menu-exec-app", "transfer 9996 XML default", "2"); top:bindAction("menu-exec-app", "transfer 9999 XML default", "3"); top:bindAction("menu-exec-app", "transfer 9991 XML default", "4"); top:bindAction("menu-exec-app", "bridge sofia/${domain}/ 888 at conference.freeswitch.org", "1"); top:bindAction("menu-exec-app", "transfer 1234*256 enum", "5"); top:bindAction("menu-sub", "demo_ivr_submenu","6"); top:bindAction("menu-exec-app", "transfer $1 XML features", "/^(10[01][0-9])$/"); top:bindAction("menu-top", "demo_ivr_lua","9"); -- This hash defines the main IVR sub-menu. It is equivalent to the lines in ivr.conf.xml ivr_sub_def = { ["main"] = undef, ["name"] = "demo_ivr_submenu_lua", ["greet_long"] = "phrase:demo_ivr_sub_menu", ["greet_short"] = "phrase:demo_ivr_main_sub_menu_short", ["invalid_sound"] = "ivr/ivr-that_was_an_invalid_entry.wav", ["exit_sound"] = "voicemail/vm-goodbye.wav", ["confirm_macro"] = "", ["confirm_key"] = "", ["tts_engine"] = "flite", ["tts_voice"] = "rms", ["confirm_attempts"] = "3", ["inter_digit_timeout"] = "2000", ["digit_len"] = "4", ["timeout"] = "15000", ["max_failures"] = "3", ["max_timeouts"] = "2" } -- sub_menu is an object of class IVRMenu -- pass in all 16 args to the constructor to define a new IVRMenu object sub_menu = freeswitch.IVRMenu( ivr_sub_def["main"], ivr_sub_def["name"], ivr_sub_def["greet_long"], ivr_sub_def["greet_short"], ivr_sub_def["invalid_sound"], ivr_sub_def["exit_sound"], ivr_sub_def["confirm_macro"], ivr_sub_def["confirm_key"], ivr_sub_def["tts_engine"], ivr_sub_def["tts_voice"], ivr_sub_def["confirm_attempts"], ivr_sub_def["inter_digit_timeout"], ivr_sub_def["digit_len"], ivr_sub_def["timeout"], ivr_sub_def["max_failures"], ivr_sub_def["max_timeouts"] ); -- Bind the action "menu-top" to the * key sub_menu:bindAction("menu-top","demo_ivr_lua","*"); --sub_menu:execute(session,"demo_ivr_submenu_lua"); -- Run the main menu top:execute(session, "demo_ivr_lua"); -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/958d66a6/attachment.html From msc at freeswitch.org Mon Dec 14 13:15:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 13:15:26 -0800 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> Message-ID: <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: > I'm playing with demo IVR from FreeSwitch distribution and have a > problem with language settings. I would like to use Russian as a > default language for voice messages so I set in vars.xml > > > > and installed Russian sound files. It works almost correctly: all > phrases are played in Russian, but not explicitly specified .wav > files; say for > > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > > I have > > 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File > [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] > [System error : No such file or directory.] > > How to fix this and make it use the correct language? > > What about this in vars.xml? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/2cd240a4/attachment.html From anthony.minessale at gmail.com Mon Dec 14 13:34:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 15:34:48 -0600 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> Message-ID: <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> you need to get the fd number of stdin however you do it in sdp and pass it as the constructor to the esl obj On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: > Dear All, > Now i use php for ESL outbound. i get variable from stdin and > process. (i use xinetd for handle socket) > $in = fopen("php://stdin", "r"); > Problem is when i use read command for get input from DTMF. i > can't get variable. So now i use 2 php script. and use read appliction in > XML DIalplan for solve this problem. > I plan to use php handle socket like a perl in > http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl > But i want to know how PHP work like this example ? > > my $host = $new_sock->sockhost(); > > my $fd = fileno($new_sock); > my $con = new ESL::ESLconnection($fd); > > my $info = $con->getInfo(); > > > Can someoue help me ? > > > Best Regards. > > Dome C. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/d3805515/attachment-0001.html From oscav at hotmail.fr Mon Dec 14 14:01:22 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 14 Dec 2009 14:01:22 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Message-ID: <26780206.post@talk.nabble.com> Hi Anthony, What kind of software?? Is there any related licensing cost? Will it be also available for windows ?? Regards, Oscav Anthony Minessale-2 wrote: > > Software G729 will be available by the end of the month. > As for, G723 we are not currently working on it. > > > On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: > >> >> Hi, >> >> What are the solutions to support the G729/G723 codec within FreeSwitch ? >> >> Thanks >> >> >> -- >> View this message in context: >> http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26780206.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dfansler at dv-fansler.com Mon Dec 14 14:13:17 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Mon, 14 Dec 2009 14:13:17 -0800 (PST) Subject: [Freeswitch-users] Freeswitch Documentation Message-ID: <29589571.1260828798164.JavaMail.root@whwamui-apprise.pas.sa.earthlink.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/7f1e9248/attachment.html From help at pdscc.com Mon Dec 14 18:01:29 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 14 Dec 2009 18:01:29 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, Message-ID: <20091215020132.1AD1C1DB501@sinclaire.sibble.net> Hmm, I emailed the zfoneproject folks about an hour ago asking about a release date for zfone3 and was surprised about a half hour later with a call from PRZ himself. Here's what I got from the call 1) the currently released version of zfone already has support for secure pbx enrollment 2) the tivi softphone client which I am using on windows mobile and symbian smartphones does not yet have secure pbx enrollment support I contacted Tivi support and the 2.0.7 client with support for secure pbx enrollment is due out close to the end of the year, depending on various factors yada, yada, yada. Am I correct in assuming that connecting via a softphone (eikga) on a windows machine also running the latest official zfone client, and calling 9787 should give me more than just a message saying the following? 1) call is secure 2) welcome to the zrtp enrollment agent 3) thank you for calling (about 1-2 seconds after item 2) then I get a few beeps, I see the zhone client saying security is interrupted and the call drops. This is the same class of behaviour I get with the Tivi clients. On 23 Aug 2009 at 17:09, Brian West wrote: > This is because you didn't install the zrtpagent.lua script and dial > zrtp on your keypad to enroll the FS box as a trusted man in the > middle... which btw will only work with the unreleased zfone3. > > /b > > On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote: > > > I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp > > capable > > endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone > > client, > > 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i > > (symbian > > s60) and an O2 Xda Flame (windows mobile 5). > > > > All 3 endpoints are registered with FS using the default extensions > > of 1000- > > 1003 > > > > With global_setvar zrtp_secure_media=true the zrtp negotiation > > between end > > points happens but the SAS never matches,below is console output for > > a call > > between 2 of the endpoints > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From brian at freeswitch.org Mon Dec 14 18:50:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 20:50:46 -0600 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <20091215020132.1AD1C1DB501@sinclaire.sibble.net> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net> Message-ID: if you don't have ZRTP compiled in as per the wiki it won't work... their are a few changes coming to this code soon. /b On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote: > Hmm, I emailed the zfoneproject folks about an hour ago asking about a > release date for zfone3 and was surprised about a half hour later > with a call > from PRZ himself. > > Here's what I got from the call > > 1) the currently released version of zfone already has support for > secure pbx > enrollment > > 2) the tivi softphone client which I am using on windows mobile and > symbian > smartphones does not yet have secure pbx enrollment support > > I contacted Tivi support and the 2.0.7 client with support for > secure pbx > enrollment is due out close to the end of the year, depending on > various > factors yada, yada, yada. > > Am I correct in assuming that connecting via a softphone (eikga) on > a windows > machine also running the latest official zfone client, and calling > 9787 > should give me more than just a message saying the following? > > 1) call is secure > 2) welcome to the zrtp enrollment agent > 3) thank you for calling (about 1-2 seconds after item 2) > > then I get a few beeps, I see the zhone client saying security is > interrupted > and the call drops. > > This is the same class of behaviour I get with the Tivi clients. From lloyd.aloysius at gmail.com Mon Dec 14 19:06:35 2009 From: lloyd.aloysius at gmail.com (Aloysius Thevarajah Lloyd) Date: Mon, 14 Dec 2009 22:06:35 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. Message-ID: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Hi All, I am trying to setup a conference room with pin number authentication. I could not find any wiki documents. If some one help me that would be helpful. Thank you in advance. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/1711058e/attachment.html From mrene_lists at avgs.ca Mon Dec 14 19:11:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 14 Dec 2009 22:11:32 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. In-Reply-To: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> References: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Message-ID: <877D61E3-4650-4CCD-9E02-FC4F7D523056@avgs.ca> http://wiki.freeswitch.org/wiki/Mod_conference Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 14-Dec-09, at 10:06 PM, Aloysius Thevarajah Lloyd wrote: > Hi All, > > I am trying to setup a conference room with pin number > authentication. I could not find any wiki documents. If some one > help me that would be helpful. > > Thank you in advance. > > > Thanks > Lloyd > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From help at pdscc.com Mon Dec 14 19:38:45 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 14 Dec 2009 19:38:45 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, Message-ID: <20091215033848.1668A1651@sinclaire.sibble.net> I do have it compiled in as per the wiki ;-) That's why I am continually scratching my head on why it's not working. I did have to use the scrooge codec setup otherwise the enrollment messages would play at about 1/10 normal speed. :-( On 14 Dec 2009 at 20:50, Brian West wrote: > if you don't have ZRTP compiled in as per the wiki it won't work... > their are a few changes coming to this code soon. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From nagalenoj at gmail.com Mon Dec 14 20:05:45 2009 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 15 Dec 2009 09:35:45 +0530 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: Great job. You would hear lots of thanks, if you could hear the entire world.! Thanks. Regards, Nagalenoj H. On Mon, Dec 14, 2009 at 6:57 AM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Robert > Clayton > *Sent:* Saturday, December 12, 2009 10:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch Documentation > > > > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler < > dfansler at dv-fansler.com> wrote: > > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/efbfc88a/attachment-0001.html From dmitry.bely at gmail.com Tue Dec 15 02:58:08 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Tue, 15 Dec 2009 13:58:08 +0300 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> Message-ID: <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote: > > > On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: >> >> I'm playing with demo IVR from FreeSwitch distribution and have a >> problem with language settings. I would like to use Russian as a >> default language for voice messages so I set in vars.xml >> >> ? >> >> and installed Russian sound files. It works almost correctly: all >> phrases are played in Russian, but not explicitly specified .wav >> files; say for >> >> ? ?> ? ? ? ?invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >> >> I have >> >> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File >> >> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] >> [System error : No such file or directory.] >> >> How to fix this and make it use the correct language? >> > What about this in vars.xml? > > data="sound_prefix=$${base_dir}/sounds/en/us/callie"/> Yes, that does the job. Thank you! But it looks a bit inconsistent. Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml. Why duplicate the settings? And another problem is that you cannot easily switch the language for your voice menu. - Dmitry Bely From codecomplete at free.fr Tue Dec 15 03:22:22 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:22:22 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? Message-ID: <26793254.post@talk.nabble.com> Hello I'm about to test Freeswitch on Windows, and would like to make sure I get it right: http://files.freeswitch.org/windows_installer/ contains "freeswitch-1.0.4.exe" from 03-Sep-2009 and "freeswitch.exe" from 07-Dec-2009. Am I correct in understanding that I should run the former, and then manually replace freeswitch.exe with the latest and greatest? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26793254.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 03:22:39 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:22:39 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> <26779926.post@talk.nabble.com> <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> Message-ID: <26793258.post@talk.nabble.com> Are you typing up something for posterity's sake? If so let me know. I'll be happy to proof-read the finished product and offer suggestions. Thanks much for the clarification. I'm used to writing short documentation when I learn a new tool, so 1) it helps me understand how it works, 2) I can perform a new install faster, and 3) it helps newbies get a head-start. So in short: - profiles = User Agents ("end points of a phone call", says Wikipedia) where each profile listens on a given IP/port so that a single Freeswitch server can handle several profiles concurrently, eg. one end-point for internal, authenticated users, and another end-point for incoming calls from a VoIP provider - contexts = tells what a caller can do; a call can go through multiple contexts during the length of the call. In a dialplan, contexts = groups of extensions Thank you. -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26793258.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 03:38:11 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:38:11 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? In-Reply-To: <26793254.post@talk.nabble.com> References: <26793254.post@talk.nabble.com> Message-ID: <26793434.post@talk.nabble.com> I'll answer my own question ;-) Freeswitch.exe = "svn 15826", ie. a temporary installer until 1.0.5 is available. -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26793434.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 04:32:19 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 04:32:19 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? In-Reply-To: <26793434.post@talk.nabble.com> References: <26793254.post@talk.nabble.com> <26793434.post@talk.nabble.com> Message-ID: <26794035.post@talk.nabble.com> I have a suggestion to make for the Windows installer: Make installing FreePBX/FusionPBX an option, as it adds 100MB although not everyone wants a web GUI to manage Freeswitch. My .15? -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26794035.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ovvenkatesan at gmail.com Tue Dec 15 05:36:42 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 15 Dec 2009 19:06:42 +0530 Subject: [Freeswitch-users] hardware requirement to run voip application Message-ID: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Hi to all, I dont know, whether I can post this question here or not, I dont have any other options . I hope some one will help me here. Here is my question? 1. I have developed my voip application on top of freeSwitch, Its working fine with soft phone. 2. Now, I need to test my application by calling my landline. I am living in India. I did some googling, I got to this hardware, *Linksys SPA3102 *. Is this only hardware enough to run my voip application or need more hardware? I am very new to this platform, and not having knowledge on hardware. Can anyone please suggest me which hardware which is freeswitch friendly? -- Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/637142cb/attachment.html From dome at tel.co.th Tue Dec 15 05:38:13 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 15 Dec 2009 20:38:13 +0700 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> Message-ID: <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> 2009/12/15 Anthony Minessale > you need to get the fd number of stdin however you do it in sdp and pass it > as the constructor to the esl obj > > It's work. Thanks. but i found PHP not good for this case. PHP need more resource. LUA look better. Now i'm testing by mod_lua but i plan to mover LUA work with outbound socket. but not found about lua outbounf socket in WIKI Best Regards. Dome C. > > > On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: > >> Dear All, >> Now i use php for ESL outbound. i get variable from stdin and >> process. (i use xinetd for handle socket) >> $in = fopen("php://stdin", "r"); >> Problem is when i use read command for get input from DTMF. i >> can't get variable. So now i use 2 php script. and use read appliction in >> XML DIalplan for solve this problem. >> I plan to use php handle socket like a perl in >> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl >> But i want to know how PHP work like this example ? >> >> my $host = $new_sock->sockhost(); >> >> my $fd = fileno($new_sock); >> my $con = new ESL::ESLconnection($fd); >> >> my $info = $con->getInfo(); >> >> >> Can someoue help me ? >> >> >> Best Regards. >> >> Dome C. >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/7339352f/attachment.html From stevendt at primrosebank.net Tue Dec 15 06:35:32 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 15 Dec 2009 14:35:32 -0000 Subject: [Freeswitch-users] hardware requirement to run voip application References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Message-ID: Hi, I am almost as new to all this as you, but, here is the benefit of my "experience" ! I cannot suggest any alternative hardware for you, although I'm sure others will, but can confirm that the SPA3102 is "FreeSwitch Friendly" - or rather, FreeSwitch is SPA3102 Friendly ! I started off with Softphones too, then added a few SIP hardware phones. With FreeSwitch installed and working locally, I then added the SPA3102 to allow me to dial out on the PSTN line. The SPA3102 was fairly straightforward to setup, but see the Wiki for the Linksys bug on RTP packet size that you'll need to adjust in the SPA3102 setup http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo regards Dave ----- Original Message ----- From: ovvenkat To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, December 15, 2009 1:36 PM Subject: [Freeswitch-users] hardware requirement to run voip application Hi to all, I dont know, whether I can post this question here or not, I dont have any other options . I hope some one will help me here. Here is my question? 1. I have developed my voip application on top of freeSwitch, Its working fine with soft phone. 2. Now, I need to test my application by calling my landline. I am living in India. I did some googling, I got to this hardware, Linksys SPA3102 . Is this only hardware enough to run my voip application or need more hardware? I am very new to this platform, and not having knowledge on hardware. Can anyone please suggest me which hardware which is freeswitch friendly? -- Regards Venkatesan OV. ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e1a0c849/attachment-0001.html From anthony.minessale at gmail.com Tue Dec 15 06:49:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Dec 2009 08:49:11 -0600 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> Message-ID: <191c3a030912150649m15e1cbfdkbb11ab67dc48d55e@mail.gmail.com> there is a lua esl wrapper too the are all based on the same C code for the ESL obj. same rule applies get file number of stdin and pass to constructor. I like perl the best for ESL but that's just me. On Tue, Dec 15, 2009 at 7:38 AM, Dome Charoenyost wrote: > > > 2009/12/15 Anthony Minessale > > you need to get the fd number of stdin however you do it in sdp and pass it >> as the constructor to the esl obj >> >> It's work. Thanks. but i found PHP not good for this case. PHP need more > resource. LUA look better. > Now i'm testing by mod_lua but i plan to mover LUA work with outbound > socket. but not found about lua outbounf socket in WIKI > > > Best Regards. > > Dome C. > > > > >> >> >> On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: >> >>> Dear All, >>> Now i use php for ESL outbound. i get variable from stdin and >>> process. (i use xinetd for handle socket) >>> $in = fopen("php://stdin", "r"); >>> Problem is when i use read command for get input from DTMF. i >>> can't get variable. So now i use 2 php script. and use read appliction in >>> XML DIalplan for solve this problem. >>> I plan to use php handle socket like a perl in >>> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl >>> But i want to know how PHP work like this example ? >>> >>> my $host = $new_sock->sockhost(); >>> >>> my $fd = fileno($new_sock); >>> my $con = new ESL::ESLconnection($fd); >>> >>> my $info = $con->getInfo(); >>> >>> >>> Can someoue help me ? >>> >>> >>> Best Regards. >>> >>> Dome C. >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/b9aaedc6/attachment.html From ovvenkatesan at gmail.com Tue Dec 15 07:03:40 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 15 Dec 2009 20:33:40 +0530 Subject: [Freeswitch-users] hardware requirement to run voip application In-Reply-To: References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Message-ID: <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> Hi Dave, Thank you very much for your quick reply and suggestion. I will try with SPA3102 then. Thanks again. Venkat. On Tue, Dec 15, 2009 at 8:05 PM, Dave Stevenson wrote: > Hi, > > I am almost as new to all this as you, but, here is the benefit of my > "experience" ! > > I cannot suggest any alternative hardware for you, although I'm sure others > will, but can confirm that the SPA3102 is "FreeSwitch Friendly" - or rather, > FreeSwitch is SPA3102 Friendly ! > > I started off with Softphones too, then added a few SIP hardware phones. > > With FreeSwitch installed and working locally, I then added the SPA3102 to > allow me to dial out on the PSTN line. > > The SPA3102 was fairly straightforward to setup, but see the Wiki for the > Linksys bug on RTP packet size that you'll need to adjust in the SPA3102 > setup > > http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo > > regards > Dave > > ----- Original Message ----- > *From:* ovvenkat > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, December 15, 2009 1:36 PM > *Subject:* [Freeswitch-users] hardware requirement to run voip application > > Hi to all, > > I dont know, whether I can post this question here or not, I dont have any > other options . I hope some one will help me here. Here is my question? > > 1. I have developed my voip application on top of freeSwitch, Its working > fine with soft phone. > 2. Now, I need to test my application by calling my landline. I am living > in India. > > I did some googling, I got to this hardware, *Linksys SPA3102 *. Is this > only hardware enough to run my voip application or need more hardware? > > I am very new to this platform, and not having knowledge on hardware. Can > anyone please suggest me which hardware which is freeswitch friendly? > > -- > > Regards > Venkatesan OV. > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e08feff6/attachment.html From nicolas at medularis.com Tue Dec 15 08:22:29 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 13:22:29 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? Message-ID: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> I'm looking for the equivalent configuration parameter or option of Asterisk's canreinvite (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is there anything like this for configuring a gateway? (there's no info about it on the wiki). Thanks! Nicolas From kristian.kielhofner at gmail.com Tue Dec 15 08:45:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 15 Dec 2009 11:45:01 -0500 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> Message-ID: <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> Closest thing I've found: http://wiki.freeswitch.org/wiki/Channel_Variables#bypass_media_after_bridge On Tue, Dec 15, 2009 at 11:22 AM, Nicolas Brenner wrote: > I'm looking for the equivalent configuration parameter or option of > Asterisk's canreinvite > (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is > there anything like this for configuring a gateway? (there's no info > about it on the wiki). > > Thanks! > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From oscav at hotmail.fr Tue Dec 15 09:24:19 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 15 Dec 2009 09:24:19 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26798406.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26798406.post@talk.nabble.com> Message-ID: <26798443.post@talk.nabble.com> Howler modules are available only for unix platforms. I'm running FS on windows. PBriffett wrote: > > Howler Technologies have a FS compliant G.729 codec and hardware solution. > Feel free to go and get the free trial > > Anthony Minessale-2 wrote: >> >> Software G729 will be available by the end of the month. >> As for, G723 we are not currently working on it. >> >> >> On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: >> >>> >>> Hi, >>> >>> What are the solutions to support the G729/G723 codec within FreeSwitch >>> ? >>> >>> Thanks >>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26798443.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From oscav at hotmail.fr Tue Dec 15 09:27:03 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 15 Dec 2009 09:27:03 -0800 (PST) Subject: [Freeswitch-users] stream G729 RTP payload in passthrough Message-ID: <26798489.post@talk.nabble.com> Hi, Would it be possible to "play" a file that is a RTP payload saved from Wireshark, in order to use the G729 passthrough while playing files to caller?? Thanks -- View this message in context: http://old.nabble.com/stream-G729-RTP-payload-in-passthrough-tp26798489p26798489.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Tue Dec 15 09:54:39 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 15 Dec 2009 09:54:39 -0800 Subject: [Freeswitch-users] One-way Video Message-ID: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> I am trying to bring up a video call, but not having much luck. We are only getting one-way video (i.e. the caller sees far-end video, but the callee does not). I added the H263/H264 tags to the pre-process "global_codec_prefs" and "outbound_codec_prefs" tags in vars.xml. Anyone have hints on making two-way video to work? Best Regards, Jerry From nicolas at medularis.com Tue Dec 15 10:00:19 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 15:00:19 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> Message-ID: <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> Thanks, but I would like to keep FS in the media path. What would be the equivalent of an Asterisk sip.conf's canreinvite=no? On Tue, Dec 15, 2009 at 1:45 PM, Kristian Kielhofner wrote: > Closest thing I've found: > > http://wiki.freeswitch.org/wiki/Channel_Variables#bypass_media_after_bridge > From frank at carmickle.com Tue Dec 15 10:12:54 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 15 Dec 2009 13:12:54 -0500 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> Message-ID: <20091215181254.GA31924@base.carmickle.com> On Tue, Dec 15, Nicolas Brenner wrote: > Thanks, but I would like to keep FS in the media path. What would be > the equivalent of an Asterisk sip.conf's canreinvite=no? It's that way by default. Fs wants to listen for events on a channel in the default config. See bind_meta_app. --FC From nicolas at medularis.com Tue Dec 15 10:26:40 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 15:26:40 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <20091215181254.GA31924@base.carmickle.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> <20091215181254.GA31924@base.carmickle.com> Message-ID: <1b46b4e80912151026x22c8a1b4xb639cd23e2835ce9@mail.gmail.com> Thanks! On Tue, Dec 15, 2009 at 3:12 PM, Frank Carmickle wrote: > On Tue, Dec 15, Nicolas Brenner wrote: >> Thanks, but I would like to keep FS in the media path. What would be >> the equivalent of an Asterisk sip.conf's canreinvite=no? > > It's that way by default. ?Fs wants to listen for events on a channel in the default config. ?See bind_meta_app. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jarrod at fed-com.com Mon Dec 14 20:55:44 2009 From: jarrod at fed-com.com (Jarrod Lash) Date: Mon, 14 Dec 2009 23:55:44 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. In-Reply-To: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> References: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Message-ID: <8c2388d80912142055y675f245aib2d54903a15ac05@mail.gmail.com> Lloyd, I used this sometime ago to setup ours... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR -- Jarrod Lash, Federated Communications, LLC. www.fed-com.com Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368 On Mon, Dec 14, 2009 at 10:06 PM, Aloysius Thevarajah Lloyd wrote: > Hi All, > > I am trying to setup a conference room with pin number authentication. I > could not find any wiki documents. If some one help me that would be > helpful. > > Thank you in advance. > > > Thanks > Lloyd > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevesteffler at shaw.ca Tue Dec 15 08:04:37 2009 From: stevesteffler at shaw.ca (Steve Steffler) Date: Tue, 15 Dec 2009 09:04:37 -0700 Subject: [Freeswitch-users] mod_voicemail question Message-ID: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> Hi all, What is the difference between the mod_voicemail "vm_message_ext" parameter and the "file-extension" parameter? I want all my voicemail in .WAV format except for a couple of extensions which need to be in MP3. I'm getting strange results playing with these settings, for example, after logging into the voicemail, it will say "You have 1 new message. First message at ", and then instead of the voicemail message there will be silence and a long pause. Then it will repeat the message count and loop this behavior. During the silence, I seem to be able to press keys to trigger voicemail events, like for example I am allowed to delete the message (although it isn't playing the message to me, and I am instead hearing silence). Any ideas? Steve From kristian.kielhofner at gmail.com Tue Dec 15 11:06:19 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 15 Dec 2009 14:06:19 -0500 Subject: [Freeswitch-users] stream G729 RTP payload in passthrough In-Reply-To: <26798489.post@talk.nabble.com> References: <26798489.post@talk.nabble.com> Message-ID: <2d9149cd0912151106u696e79c2n12dec12051d9f4b5@mail.gmail.com> If your file is G729 on disk you can play it with mod_native_file. No need to deal with RTP, pcaps, etc. On Tue, Dec 15, 2009 at 12:27 PM, Oscav wrote: > > Hi, > > Would it be possible to "play" a file that is a RTP payload saved from > Wireshark, in order to use the G729 passthrough while playing files to > caller?? > > Thanks -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Tue Dec 15 11:09:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 11:09:09 -0800 Subject: [Freeswitch-users] mod_voicemail question In-Reply-To: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> References: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> Message-ID: <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> On Tue, Dec 15, 2009 at 8:04 AM, Steve Steffler wrote: > > Hi all, > > What is the difference between the mod_voicemail "vm_message_ext" parameter > and the "file-extension" parameter? > vm_message_ext is a channel variable: http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext file-extension is a parameter of the voicemail module: http://wiki.freeswitch.org/wiki/Mod_voicemail#file-extension The former sets for a specific user, the latter for mod_voicemail in general. > > I want all my voicemail in .WAV format except for a couple of extensions > which need to be in MP3. > > I'm getting strange results playing with these settings, for example, after > logging into the voicemail, it will say "You have 1 new message. First > message at ", and then instead of the voicemail message there > will be silence and a long pause. Then it will repeat the message count and > loop this behavior. During the silence, I seem to be able to press keys to > trigger voicemail events, like for example I am allowed to delete the > message (although it isn't playing the message to me, and I am instead > hearing silence). > > Any ideas? > Is this perhaps a recording of silence, so that you might actually be listening to a message? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/d5af3742/attachment.html From msc at freeswitch.org Tue Dec 15 11:35:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 11:35:45 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Dec 18 - Need Your Items Message-ID: <87f2f3b90912151135m4677a553k397b36369a924add@mail.gmail.com> Hello friends, Just to let you know, I have posted the FS weekly conf call agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 It's pretty clean at this point so if you've got things that you'd like to discuss with the group then please add your items to the list. If you have items that require attention, like documentation and janitorial items then by all means drop those on the list as well. One thing we do need to discuss is how we will accomplish screen casting. We are going to have presentations and we want as many people as possible to be able to view the screen while listening to the conference. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/81c8634a/attachment-0001.html From malay.thakershi at continuityhealth.com Tue Dec 15 11:56:41 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 15 Dec 2009 13:56:41 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files Message-ID: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> Hello, I create one WAV file that has: Question + Option 1 + Option 2 + Option 3 + . I noticed towards end of the file Cepstral Allison starts chopping and speeding up. So my question text that gets converted to WAV file using swift EXE looks like: Which is the biggest mammal on land? Select one of the following choices.Or press star to skip the question 1 Parrot 2 Elephant 3 T-Rex 4 Blue Whale And my csharp code looks like: pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", "^\\d", ""); What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? Please help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/d5a8ef7f/attachment.html From lists at redbonez.net Tue Dec 15 12:03:11 2009 From: lists at redbonez.net (Adam Ford) Date: Tue, 15 Dec 2009 13:03:11 -0700 Subject: [Freeswitch-users] Caller ID issue Message-ID: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> I have been having a problem with my outgoing caller ID coming through as Private or Restricted using a PRI + Openzap. My provider claims that it must be a configuration on my end. Is there something I might be missing? Setup is essential the default FreeSWITCH configuration. I realize this is pretty vague, but I don't really know where to start to troubleshoot this issue on my end. Caller ID works from SIP UA to SIP UA as well as for all incoming calls. I don't even know where to configure FreeSWITCH to mask my caller ID as private or restricted. Thank-AF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/f4aea3b1/attachment.html From bcxml at hotmail.com Tue Dec 15 12:11:37 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 12:11:37 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 Message-ID: <26801000.post@talk.nabble.com> I have Freeswitch and Microsoft Speech Server 2007 on the same box When Speech Server initiates a call, I get a sip error message 480 Here is the internal profile trace... freeswitch at HD-T2253CN> freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011: ------------------------------------------------------------------------ INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 FROM: ;epid=55D003BB53;tag=25bf 436a29 TO: CSEQ: 2 INVITE CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTACT: ;automata CONTENT-LENGTH: 340 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 209.172.55.154 s=Microsoft Speech Server session c=IN IP4 209.172.55.154 t=0 0 m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: ;epid=55D003BB53;tag=25bf 436a29 TO: CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Content-Length: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel sofia/inter nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing 12482578002- >19059183027 in context public 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /internal/12482578002 at 127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: ;epid=55D003BB53;tag=25bf 436a29 To: ;tag=gr4aF6aS8tZ0j CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "19059183027" ------------------------------------------------------------------------ recv 383 bytes from tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ ACK sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 FROM: ;tag=25bf436a29;epid=55D0 03BB53 TO: ;tag=gr4aF6aS8tZ0j CSEQ: 2 ACK CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTENT-LENGTH: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/ internal/12482578002 at 127.0.0.1:5080) Ended 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1156 Close Channel sof ia/internal/12482578002 at 127.0.0.1:5080 [CS_DESTROY] Can anyone point me in the right direction ? Thanks Brian -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26801000.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lists at redbonez.net Tue Dec 15 12:50:54 2009 From: lists at redbonez.net (Adam Ford) Date: Tue, 15 Dec 2009 13:50:54 -0700 Subject: [Freeswitch-users] Caller ID issue In-Reply-To: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> References: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> Message-ID: <00a601ca7dc8$4cfd99a0$e6f8cce0$@net> Scratch that, I had my Openzap configured for national, not NI2. Thanks, -AF From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Ford Sent: Tuesday, December 15, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Caller ID issue I have been having a problem with my outgoing caller ID coming through as Private or Restricted using a PRI + Openzap. My provider claims that it must be a configuration on my end. Is there something I might be missing? Setup is essential the default FreeSWITCH configuration. I realize this is pretty vague, but I don't really know where to start to troubleshoot this issue on my end. Caller ID works from SIP UA to SIP UA as well as for all incoming calls. I don't even know where to configure FreeSWITCH to mask my caller ID as private or restricted. Thank-AF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a16f1f7c/attachment.html From msc at freeswitch.org Tue Dec 15 13:27:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 13:27:12 -0800 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26801000.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> Message-ID: <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> On Tue, Dec 15, 2009 at 12:11 PM, bcxml wrote: > > I have Freeswitch and Microsoft Speech Server 2007 on the same box > > When Speech Server initiates a call, I get a sip error message 480 > > Here is the internal profile trace... > > freeswitch at HD-T2253CN> > > freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at > 20:04:05 > .445011: > ------------------------------------------------------------------------ > INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 > FROM: > ;epid=55D003BB53;tag=25bf > 436a29 > TO: > CSEQ: 2 INVITE > CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe > MAX-FORWARDS: 70 > VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 > CONTACT: > 704290e5b4e03b>;automata > CONTENT-LENGTH: 340 > USER-AGENT: RTCC/3.0.0.0 > CONTENT-TYPE: application/sdp > ALLOW: UPDATE > ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify > > v=0 > o=- 0 0 IN IP4 209.172.55.154 > s=Microsoft Speech Server session > c=IN IP4 209.172.55.154 > t=0 0 > m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 > a=rtpmap:114 x-msrta/16000 > a=fmtp:114 bitrate=29000 > a=rtpmap:115 x-msrta/8000 > a=fmtp:115 bitrate=11800 > a=rtpmap:97 RED/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 > FROM: > ;epid=55D003BB53;tag=25bf > 436a29 > TO: > CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe > CSEQ: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel > sofia/inter > nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] > 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing > 12482578002- > >19059183027 in context public > Are you handling "19059183027" in the public context? If so, what is that extension doing with the call? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e6e20ee3/attachment-0001.html From mike at jerris.com Tue Dec 15 13:40:14 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 16:40:14 -0500 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> Message-ID: <1B812EFC-3A6F-428E-BFEB-A2228CC1A3F8@jerris.com> The issue is the demo ivr does not use phrase macros. The line in ru.xml is for the phrase macros. We should probably change this in the future. Mike On Dec 15, 2009, at 5:58 AM, Dmitry Bely wrote: > On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote: >> >> >> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: >>> >>> I'm playing with demo IVR from FreeSwitch distribution and have a >>> problem with language settings. I would like to use Russian as a >>> default language for voice messages so I set in vars.xml >>> >>> >>> >>> and installed Russian sound files. It works almost correctly: all >>> phrases are played in Russian, but not explicitly specified .wav >>> files; say for >>> >>> >> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >>> >>> I have >>> >>> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File >>> >>> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] >>> [System error : No such file or directory.] >>> >>> How to fix this and make it use the correct language? >>> >> What about this in vars.xml? >> >> > data="sound_prefix=$${base_dir}/sounds/en/us/callie"/> > > Yes, that does the job. Thank you! But it looks a bit inconsistent. > Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml. > Why duplicate the settings? And another problem is that you cannot > easily switch the language for your voice menu. > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bcxml at hotmail.com Tue Dec 15 13:47:39 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 13:47:39 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> Message-ID: <26802352.post@talk.nabble.com> I have the following setup.... conf\dialplan\public\VoipMs.xml conf\sip_profiles\external\VoipMs.xml mercutioviz wrote: > > On Tue, Dec 15, 2009 at 12:11 PM, bcxml wrote: > >> >> I have Freeswitch and Microsoft Speech Server 2007 on the same box >> >> When Speech Server initiates a call, I get a sip error message 480 >> >> Here is the internal profile trace... >> >> freeswitch at HD-T2253CN> >> >> freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at >> 20:04:05 >> .445011: >> >> ------------------------------------------------------------------------ >> INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 >> FROM: >> ;epid=55D003BB53;tag=25bf >> 436a29 >> TO: >> CSEQ: 2 INVITE >> CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe >> MAX-FORWARDS: 70 >> VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 >> CONTACT: >> > 704290e5b4e03b>;automata >> CONTENT-LENGTH: 340 >> USER-AGENT: RTCC/3.0.0.0 >> CONTENT-TYPE: application/sdp >> ALLOW: UPDATE >> ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify >> >> v=0 >> o=- 0 0 IN IP4 209.172.55.154 >> s=Microsoft Speech Server session >> c=IN IP4 209.172.55.154 >> t=0 0 >> m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 >> a=rtpmap:114 x-msrta/16000 >> a=fmtp:114 bitrate=29000 >> a=rtpmap:115 x-msrta/8000 >> a=fmtp:115 bitrate=11800 >> a=rtpmap:97 RED/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 >> FROM: >> ;epid=55D003BB53;tag=25bf >> 436a29 >> TO: >> CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe >> CSEQ: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel >> sofia/inter >> nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] >> 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing >> 12482578002- >> >19059183027 in context public >> > > Are you handling "19059183027" in the public context? If so, what is that > extension doing with the call? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26802352.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Dec 15 14:01:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:01:13 -0500 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26780206.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> Message-ID: <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> We have not published costs yet, but expect it to be inline with other similar offerings. I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives but we have not yet tested the code on windows. Mike On Dec 14, 2009, at 5:01 PM, Oscav wrote: > > Hi Anthony, > > What kind of software?? Is there any related licensing cost? Will it be also > available for windows ?? From a.alalousi at gmail.com Tue Dec 15 14:00:28 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 15 Dec 2009 22:00:28 +0000 Subject: [Freeswitch-users] REDIRECT 503 not working Message-ID: People, I have a very simple call scenario where calls are hitting FreeSWITCH, and I need to send a 302 REDIRECT to get them to go elsewhere without answering them. It's for a phased migration requirement so that traffic can continue to flow to the current site, but gets redirected to a new site. The old site will eventually be decommissioned. Here is what I have in my conf/dialplan/public/test.xml: FreeSWITCH is sending back the 302 back to the test end-point (eyeBeam 1.5.20 build 54436), but the call is not reaching the specified in the data portion of the redirect application. I know it's sending it because of logs FreeSWITCH end and the info being displayed on eyeBeam's client interface stating Call being forwarded ...etc. ...etc. Has anyone had any similar experiences with a similar setup ? Oh, and one more thing, I have disabled firewalling on both the proxy where eyeBeam is registered and the destination where I'm sending the call. I have also verified that my new destination (also a FreeSWITCH box) is accepting registrations, inviites and able to route calls initiated by eyeBeam when directly registered on it. Has anyone had similar experiences ? better still, has anyone successfully setup FreeSWITCH to be an SBC and can give me feedback ? Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS, CCIE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/224cb6ca/attachment.html From mike at jerris.com Tue Dec 15 14:04:07 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:04:07 -0500 Subject: [Freeswitch-users] One-way Video In-Reply-To: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> References: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> Message-ID: <79F2C126-EBF5-403F-BEC0-3B0FB287046E@jerris.com> try just 1 video codec in freeswitch codec prefs and make sure you are using trunk, we fixed quite a few video issues recently. Mike On Dec 15, 2009, at 12:54 PM, Jerry Richards wrote: > I am trying to bring up a video call, but not having much luck. We are only > getting one-way video (i.e. the caller sees far-end video, but the callee > does not). I added the H263/H264 tags to the pre-process > "global_codec_prefs" and "outbound_codec_prefs" tags in vars.xml. > > Anyone have hints on making two-way video to work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/9292db96/attachment.html From mike at jerris.com Tue Dec 15 14:05:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:05:25 -0500 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> References: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> Message-ID: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> You can do that with phrase macros. Mike On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: > Hello, I create one WAV file that has: > > Question + Option 1 + Option 2 + Option 3 + ? > > I noticed towards end of the file Cepstral Allison starts chopping and speeding up. > > So my question text that gets converted to WAV file using swift EXE looks like: > > Which is the biggest mammal on land? > Select one of the following choices.Or press star to skip the question > 1 Parrot > 2 Elephant > 3 T-Rex > 4 Blue Whale > > > And my csharp code looks like: > pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", > "^\\d", ""); > > > What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. > > I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? > > Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/bdf3d478/attachment-0001.html From mike at jerris.com Tue Dec 15 14:09:04 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:09:04 -0500 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26801000.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> Message-ID: Try turning on debug logs, but from this it looks like its not matching any extensions. Mike On Dec 15, 2009, at 3:11 PM, bcxml wrote: > > I have Freeswitch and Microsoft Speech Server 2007 on the same box > > When Speech Server initiates a call, I get a sip error message 480 > > Here is the internal profile trace... > > 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel > sofia/inter > nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] > 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing > 12482578002- >> 19059183027 in context public > 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup > sofia > /internal/12482578002 at 127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] > send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: From dule.maillist at gmail.com Tue Dec 15 14:22:53 2009 From: dule.maillist at gmail.com (Dan Le) Date: Tue, 15 Dec 2009 17:22:53 -0500 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> Message-ID: <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string Dan On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: > You can do that with phrase macros. > > Mike > > On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: > > Hello, I create one WAV file that has: > > Question + Option 1 + Option 2 + Option 3 + ? > > I noticed towards end of the file Cepstral Allison starts chopping and > speeding up. > > So my question text that gets converted to WAV file using swift EXE looks > like: > > Which is the biggest mammal on land? > Select one of the following choices. strength='weak'/>Or press star to skip the question > 1 Parrot > 2 Elephant > 3 T-Rex > 4 Blue Whale > > > And my csharp code looks like: > pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, > 5000, "*#", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV > ", > > @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", > "^\\d", ""); > > > What happens is, the voice just starts chopping and speeding up between > options. Even though I am not able to say that it only does that towards the > end, I think so. > > I thought, if I break each file into individual WAV instead of 1 big WAV, > it may help? > > Is there a way to play multiple (separate) WAV files in PlayAndGetDigits > function? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/7ec4d037/attachment.html From msc at freeswitch.org Tue Dec 15 14:54:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 14:54:52 -0800 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: References: <26801000.post@talk.nabble.com> Message-ID: <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: > Try turning on debug logs, but from this it looks like its not matching any > extensions. > > Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e759a4ed/attachment.html From brian at freeswitch.org Tue Dec 15 15:02:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Dec 2009 17:02:43 -0600 Subject: [Freeswitch-users] REDIRECT 503 not working In-Reply-To: References: Message-ID: <2D31351D-6577-47DC-B9AC-EB1BE9ACAF6B@freeswitch.org> You have to be careful things like eyebeam will send the invite back to FS1 that did the redirect as if it were the proxy with the request URI as the URI you did in the 302 please post a sip trace of the entire exchange on pastebin. /b On Dec 15, 2009, at 4:00 PM, Ahmed Naji wrote: > People, > > I have a very simple call scenario where calls are hitting > FreeSWITCH, and I need to send a 302 REDIRECT to get them to go > elsewhere without answering them. > > It's for a phased migration requirement so that traffic can continue > to flow to the current site, but gets redirected to a new site. The > old site will eventually be decommissioned. > > Here is what I have in my conf/dialplan/public/test.xml: > > > > > > > > > > FreeSWITCH is sending back the 302 back to the test end-point > (eyeBeam 1.5.20 build 54436), but the call is not reaching the > specified in the data portion of the redirect application. I know > it's sending it because of logs FreeSWITCH end and the info being > displayed on eyeBeam's client interface stating Call being > forwarded ...etc. ...etc. > > Has anyone had any similar experiences with a similar setup ? > > Oh, and one more thing, I have disabled firewalling on both the > proxy where eyeBeam is registered and the destination where I'm > sending the call. I have also verified that my new destination (also > a FreeSWITCH box) is accepting registrations, inviites and able to > route calls initiated by eyeBeam when directly registered on it. > > Has anyone had similar experiences ? better still, has anyone > successfully setup FreeSWITCH to be an SBC and can give me feedback ? > > Regards, > > Ahmed. > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS, CCIE > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Tue Dec 15 15:05:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 15:05:21 -0800 Subject: [Freeswitch-users] REDIRECT 503 not working In-Reply-To: References: Message-ID: <87f2f3b90912151505q711c34d3qda112279379a9843@mail.gmail.com> Can you turn on debug and sip trace and pastebin the console output? Reply to this thread with the pastebin URL... I'm sure some of the networking gurus can help. -MC On Tue, Dec 15, 2009 at 2:00 PM, Ahmed Naji wrote: > People, > > I have a very simple call scenario where calls are hitting FreeSWITCH, and > I need to send a 302 REDIRECT to get them to go elsewhere without answering > them. > > It's for a phased migration requirement so that traffic can continue to > flow to the current site, but gets redirected to a new site. The old site > will eventually be decommissioned. > > Here is what I have in my conf/dialplan/public/test.xml: > > > > expression="^(?:7153)(\d+)$"> > data="sip:7153$1 at aaa.bbb.ccc.ddd"/> > > > > > FreeSWITCH is sending back the 302 back to the test end-point (eyeBeam > 1.5.20 build 54436), but the call is not reaching the specified in the data > portion of the redirect application. I know it's sending it because of logs > FreeSWITCH end and the info being displayed on eyeBeam's client interface > stating Call being forwarded ...etc. ...etc. > > Has anyone had any similar experiences with a similar setup ? > > Oh, and one more thing, I have disabled firewalling on both the proxy where > eyeBeam is registered and the destination where I'm sending the call. I have > also verified that my new destination (also a FreeSWITCH box) is accepting > registrations, inviites and able to route calls initiated by eyeBeam when > directly registered on it. > > Has anyone had similar experiences ? better still, has anyone successfully > setup FreeSWITCH to be an SBC and can give me feedback ? > > Regards, > > Ahmed. > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS, CCIE > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/63966943/attachment.html From anthony.minessale at gmail.com Tue Dec 15 15:12:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Dec 2009 17:12:01 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> Message-ID: <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> make sure you are using latest trunk because it should not be sounding choppy. On Tue, Dec 15, 2009 at 4:22 PM, Dan Le wrote: > Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string > > Dan > > On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: > >> You can do that with phrase macros. >> >> Mike >> >> On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: >> >> Hello, I create one WAV file that has: >> >> Question + Option 1 + Option 2 + Option 3 + ? >> >> I noticed towards end of the file Cepstral Allison starts chopping and >> speeding up. >> >> So my question text that gets converted to WAV file using swift EXE looks >> like: >> >> Which is the biggest mammal on land? >> Select one of the following choices.> strength='weak'/>Or press star to skip the question >> 1 Parrot >> 2 Elephant >> 3 T-Rex >> 4 Blue Whale >> >> >> And my csharp code looks like: >> pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, >> 5000, "*#", >> @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV >> ", >> >> @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", >> "^\\d", ""); >> >> >> What happens is, the voice just starts chopping and speeding up between >> options. Even though I am not able to say that it only does that towards the >> end, I think so. >> >> I thought, if I break each file into individual WAV instead of 1 big WAV, >> it may help? >> >> Is there a way to play multiple (separate) WAV files in PlayAndGetDigits >> function? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/396e7ea7/attachment-0001.html From malay.thakershi at continuityhealth.com Tue Dec 15 16:26:31 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 15 Dec 2009 18:26:31 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> Message-ID: <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> Regarding Choppy issue: How do I know what trunk I am using? I had downloaded windows installer from website and installed it. Then I had to insert few DLL files from the build to get it up and running. Is it possible to only get updated files from the latest trunk? Thank you for help. Also thank "Dan" for suggesting mod_file solution for combining files for playback. Malay Thakershi From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, December 15, 2009 5:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits multiple WAV files make sure you are using latest trunk because it should not be sounding choppy. On Tue, Dec 15, 2009 at 4:22 PM, Dan Le wrote: Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string Dan On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: You can do that with phrase macros. Mike On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: Hello, I create one WAV file that has: Question + Option 1 + Option 2 + Option 3 + . I noticed towards end of the file Cepstral Allison starts chopping and speeding up. So my question text that gets converted to WAV file using swift EXE looks like: Which is the biggest mammal on land? Select one of the following choices.Or press star to skip the question 1 Parrot 2 Elephant 3 T-Rex 4 Blue Whale And my csharp code looks like: pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", "^\\d", ""); What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a937c3e3/attachment.html From brian at freeswitch.org Tue Dec 15 16:41:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Dec 2009 18:41:56 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> Message-ID: <4621A1A0-4CCB-4814-8032-8B1CB33C18D3@freeswitch.org> Compile it yourself is the best bet to get the very latests and greatest code. /b On Dec 15, 2009, at 6:26 PM, Malay Thakershi wrote: > Is it possible to only get updated files from the latest trunk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a6e18387/attachment.html From bcxml at hotmail.com Tue Dec 15 18:27:54 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 18:27:54 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> Message-ID: <26805343.post@talk.nabble.com> Here is the link to the debug log http://pastebin.freeswitch.org/11521 Brian mercutioviz wrote: > > On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: > >> Try turning on debug logs, but from this it looks like its not matching >> any >> extensions. >> >> Agreed. "console loglevel debug" at the fs cli and then make a test call, > capture output, drop into pastebin.freeswitch.org, and post the URL in > this > thread. > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Tue Dec 15 18:28:51 2009 From: senakahks at gmail.com (sky1975) Date: Tue, 15 Dec 2009 18:28:51 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill - no hangup and can call without money in database Message-ID: <1260930531708-4173535.post@n2.nabble.com> Dear Sir, I have successfully installed freeSWITCH and it works fine in passthrough mode. I installed nibblebill and it deduct money from the accounts database and it works fine. but I have two problems. 1. Calls can be initiated even though there is a minus value in accounts database 2. Calls doesn't hangup when it goes to minus values. Any answers are greatly appreciated. This is my dialplan: This is the configuration file; -- View this message in context: http://n2.nabble.com/Mod-nibblebill-no-hangup-and-can-call-without-money-in-database-tp4173535p4173535.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Dec 15 18:46:41 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 21:46:41 -0500 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26805343.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> <26805343.post@talk.nabble.com> Message-ID: <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> Yep, there is your issue.. I missed it when you pasted the extension, its a typo in your condition. Dialplan: sofia/internal/12482578002 at 127.0.0.1:5080 Regex (FAIL) [VoipMs] destination_number(19059183027) =~ /expression=/ break=on-false Notice what it is comparing there .. and notice the typo in your condition. Mike On Dec 15, 2009, at 9:27 PM, bcxml wrote: > > > Here is the link to the debug log > > http://pastebin.freeswitch.org/11521 > > > Brian > > > mercutioviz wrote: >> >> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: >> >>> Try turning on debug logs, but from this it looks like its not matching >>> any >>> extensions. >>> >>> Agreed. "console loglevel debug" at the fs cli and then make a test call, >> capture output, drop into pastebin.freeswitch.org, and post the URL in >> this >> thread. >> -MC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bcxml at hotmail.com Tue Dec 15 18:58:00 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 18:58:00 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> <26805343.post@talk.nabble.com> <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> Message-ID: <26805522.post@talk.nabble.com> Mike..thank you so much... It works fine now Brian Michael Jerris wrote: > > Yep, there is your issue.. I missed it when you pasted the extension, its > a typo in your condition. > > Dialplan: sofia/internal/12482578002 at 127.0.0.1:5080 Regex (FAIL) [VoipMs] > destination_number(19059183027) =~ /expression=/ break=on-false > > Notice what it is comparing there .. > > field="destination_number"expression="expression="^1?(\d{10})$"> > > and notice the typo in your condition. > > Mike > > On Dec 15, 2009, at 9:27 PM, bcxml wrote: > >> >> >> Here is the link to the debug log >> >> http://pastebin.freeswitch.org/11521 >> >> >> Brian >> >> >> mercutioviz wrote: >>> >>> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: >>> >>>> Try turning on debug logs, but from this it looks like its not matching >>>> any >>>> extensions. >>>> >>>> Agreed. "console loglevel debug" at the fs cli and then make a test >>>> call, >>> capture output, drop into pastebin.freeswitch.org, and post the URL in >>> this >>> thread. >>> -MC >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805522.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Tue Dec 15 20:58:46 2009 From: freeswitch at aastral.net (Bill W) Date: Tue, 15 Dec 2009 23:58:46 -0500 Subject: [Freeswitch-users] ACLs through proxy Message-ID: <4B286906.7040502@aastral.net> Hi All, I have a FreeSWITCH cluster behind an OpenSIPS proxy/load balancer, and I'd like to be able to use the auth-calls feature in my sip profile in conjunction with the parameter in the directory. In addition to running the INVITEs through the load balancer, I also need to run the REGISTERs through the load balancer because some of my endpoints are behind NAT firewalls, and therefore won't accept incoming calls from IPs other than the IP they registered to. INVITEs from the cluster going to registered endpoints are sent back through the proxy, thereby solving the NAT problem. However, having the proxy in the path effectively negates using IP based ACLS. The functionality I require is as follows: 1. Only allow registration if the endpoint IP matches it's own unique acl CIDR (specified in the directory). 2. Only accept INVITEs from endpoints that authenticate AND match the acl CIDR (again, specified in the directory). Does anyone have any recommendations on the best way to get the auth-calls functionality using an IP other than the IP of the last hop? If not, how hard would it be to add a feature to the auth-calls parameter to accept a channel variable from which to obtain the actual endpoint IP? Thanks! Bill From codecomplete at free.fr Tue Dec 15 23:38:34 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 23:38:34 -0800 (PST) Subject: [Freeswitch-users] [Windows] Stable enough for production use? Message-ID: <26807322.post@talk.nabble.com> Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 23:51:32 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 23:51:32 -0800 (PST) Subject: [Freeswitch-users] hardware requirement to run voip application In-Reply-To: <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> Message-ID: <26807453.post@talk.nabble.com> The 3102 has quite a lot of settings you can play with. Here are two useful documents: SPA-3102 Simplified Users Guide Version 1.1a http://www.jmgtechnology.com.au/spa_3102_guide.pdf Linksys ATA Administrator Guide 3.2.pdf http://www.inphonex.com/download/spa8000-ag.pdf -- View this message in context: http://old.nabble.com/hardware-requirement-to-run-voip-application-tp26795029p26807453.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Wed Dec 16 00:27:41 2009 From: senakahks at gmail.com (Senaka Amarakeerthi) Date: Wed, 16 Dec 2009 17:27:41 +0900 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database Message-ID: Dear Sir, I have successfully installed freeSWITCH and it works fine in passthrough mode. I installed nibblebill and it deduct money from the accounts database and it works fine. but I have two problems. 1. Calls can be initiated even though there is a minus value in accounts database 2. Calls doesn't hangup when it goes to minus values. Any answers are greatly appreciated. This is my dialplan: This is the configuration file; From codecomplete at free.fr Wed Dec 16 01:24:23 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 16 Dec 2009 01:24:23 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Message-ID: <26808383.post@talk.nabble.com> Hello Does someone know of a free service on the web that can check whether the UDP ports on my firewall are open after Freeswitch is up and running? ShieldsUp only scans TCP ports. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26808383.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From asobihoudai at yahoo.com Wed Dec 16 01:58:46 2009 From: asobihoudai at yahoo.com (Paul) Date: Wed, 16 Dec 2009 01:58:46 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <26808383.post@talk.nabble.com> References: <26808383.post@talk.nabble.com> Message-ID: <72220.45962.qm@web111310.mail.gq1.yahoo.com> If you have a shell on an external host, nmap will kindly do that for you quickly and without charge. ----- Original Message ---- From: Fred-145 To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 1:24:23 AM Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Hello Does someone know of a free service on the web that can check whether the UDP ports on my firewall are open after Freeswitch is up and running? ShieldsUp only scans TCP ports. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26808383.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Dec 16 04:07:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 06:07:27 -0600 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B286906.7040502@aastral.net> References: <4B286906.7040502@aastral.net> Message-ID: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> use "apply-proxy-acl" on the sofia profile. /b On Dec 15, 2009, at 10:58 PM, Bill W wrote: > > However, having the proxy in the path effectively negates using IP > based > ACLS. From juanbackson at gmail.com Wed Dec 16 04:33:26 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 16 Dec 2009 20:33:26 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels Message-ID: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Hi, I am having problem with around 1 % of the channels always get zombilized. What I want to do is to have a background thread that regularly check all the channels that have been in existance for like > 1 hr, and then check to see if there is any RTP coming in and going out. If there is no RTP, then I just hangup that channel. Does anyone know if there is anyway to do that in a freeswitch module? Which API can I use to accomplish this purpose? Alternatively, is there anyway to configure freeswitch so that it will hangup the calls where there is no media in and out for so many seconds? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/6857e9b7/attachment.html From costa.zikalala at gmail.com Wed Dec 16 04:37:38 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Wed, 16 Dec 2009 14:37:38 +0200 Subject: [Freeswitch-users] Basic Question on the Internal Profile Message-ID: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> Hi All I understand that to connect to a SIP Provider you have to (amongst other things) define a Gateway on the External Profile. But some gateways may be defined on the Internal Profile. What kind of gateways would these be and what would be their purpose as most gateways are External? Hope my question makes a bit of sense. Thanks CZ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/d81f49b8/attachment-0001.html From Prometheus001 at gmx.net Wed Dec 16 04:47:57 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 16 Dec 2009 13:47:57 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN Message-ID: <4B28D6FD.6010702@gmx.net> Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)====INVITE===>FS PSTN(A)<===TRYING===>FS FS===INVITE==>PSTN(B) FS<==TRYING===PSTN(B) FS<==RINGING==PSTN(B) PSTN(A)<==PROGRESS===FS FS<===OK======PSTN(B) FS====ACK====>PSTN(B) PSTN(A)<===OK========FS PSTN(A)====ACK======>FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter From nameer.kazzaz at gmail.com Wed Dec 16 06:01:05 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Wed, 16 Dec 2009 14:01:05 +0000 Subject: [Freeswitch-users] xml_rpc.conf Message-ID: <4B28E821.20207@gmail.com> Hi all, Can I set xml_rpc server to run on a specific interface I can set the port but not the ip address to bind to. I have a linux server with more then one interface. I don't want to use iptables to block it. Thanks Nameer From brian at freeswitch.org Wed Dec 16 06:52:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 08:52:09 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Message-ID: Why not just set rtp-timeout-sec on the sofia profile and it'll do that for you. Unless something else is going on. /b On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > Hi, > > I am having problem with around 1 % of the channels always get > zombilized. > > What I want to do is to have a background thread that regularly > check all the channels that have been in existance for like > 1 hr, > and then check to see if there is any RTP coming in and going out. > If there is no RTP, then I just hangup that channel. Does anyone > know if there is anyway to do that in a freeswitch module? Which > API can I use to accomplish this purpose? Alternatively, is there > anyway to configure freeswitch so that it will hangup the calls > where there is no media in and out for so many seconds? > > Thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From neilp at cs.stanford.edu Wed Dec 16 06:56:04 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Wed, 16 Dec 2009 20:26:04 +0530 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? -Neil On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > looks like ogg devel packages are installed but ogg lib is not? > > > On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > > > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > > unknown format. Looking through the log, I found this during startup: > > > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > > Loading module /usr/local/freeswitch/mod/mod_shout.so > > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > > ogg_sync_wrote** > > > > There don't seem to be any compile-time errors, yet I can't seem to > > eliminate this issue. Any help would be appreciated. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/a10b7eca/attachment.html From peter.olsson at visionutveckling.se Wed Dec 16 07:29:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 16 Dec 2009 16:29:28 +0100 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <26807322.post@talk.nabble.com> References: <26807322.post@talk.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper> Hi, We've been running FS in win32 in a "semi-production" environment for some time now (since version 1.0.1 - following the trunk all the time since then). We use it as both a lab environment - to distribute SIP-trunks to different PBX'es, and also for "real" endpoints for some (about 10-15) of our internal users. Over the time there have been a few issues (now many though), but from 1.0.4 and later it's been very stable for our use, and no memory leaks etc. We haven't tried analog PSTN connections though, only SIP, h323 (opal) and some (not much) use of Sangoma E1 connected to one of our PBX'es. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Fred-145 [codecomplete at free.fr] Skickat: den 16 december 2009 08:38 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] [Windows] Stable enough for production use? Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4b2891a332931788118788! From mike at jerris.com Wed Dec 16 07:44:25 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Dec 2009 10:44:25 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. Thanks Mike On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. > > I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? > > -Neil > > On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > looks like ogg devel packages are installed but ogg lib is not? > > > On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > > > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > > unknown format. Looking through the log, I found this during startup: > > > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > > Loading module /usr/local/freeswitch/mod/mod_shout.so > > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > > ogg_sync_wrote** > > > > There don't seem to be any compile-time errors, yet I can't seem to > > eliminate this issue. Any help would be appreciated. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/db02c3f0/attachment.html From john_platts at hotmail.com Wed Dec 16 07:59:01 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 16 Dec 2009 09:59:01 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial Message-ID: How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ From mike at jerris.com Wed Dec 16 08:00:41 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Dec 2009 11:00:41 -0500 Subject: [Freeswitch-users] xml_rpc.conf In-Reply-To: <4B28E821.20207@gmail.com> References: <4B28E821.20207@gmail.com> Message-ID: On Dec 16, 2009, at 9:01 AM, Nameer Kazzaz wrote: > Hi all, > Can I set xml_rpc server to run on a specific interface I can set > the port but not the ip address to bind to. I have a linux server with > more then one interface. I don't want to use iptables to block it. No, but you always have the option of using iptables to block it. Mike From jpitcher at nuvio.com Wed Dec 16 08:11:05 2009 From: jpitcher at nuvio.com (Jonathan Pitcher) Date: Wed, 16 Dec 2009 08:11:05 -0800 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: Message-ID: John, To do a click to call in FS you need to have some app that connects to the ESL or Event Socket Layer and runs one of the calls diagramed here ... http://wiki.freeswitch.org/wiki/Mod_commands#originate For use with the ESL just prepend api in front of the originate so your call looks something like: $command = 'api originate user at domain &bridge(user at domain)'; As for programs able to do that from a Microsoft Product, That I am not sure of. Jonathan Pitcher On 12/16/09 9:59 AM, "John Platts" wrote: How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/9d850f19/attachment-0001.html From brian at freeswitch.org Wed Dec 16 08:18:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 10:18:28 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: Message-ID: <54F485C1-2FD2-4950-8617-C4C2F20717F9@freeswitch.org> see scripts/perl/call.cgi /b On Dec 16, 2009, at 9:59 AM, John Platts wrote: > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? > > _________________________________________________________________ > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From john_platts at hotmail.com Wed Dec 16 08:24:07 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 16 Dec 2009 10:24:07 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: , Message-ID: You've made my day. ________________________________ > From: jpitcher at nuvio.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 16 Dec 2009 08:11:05 -0800 > Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial > > > > > > > > > John, > > > > To do a click to call in FS you need to have some app that connects to the ESL or Event Socket Layer and runs one of the calls diagramed here ... > > > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > > > For use with the ESL just prepend api in front of the originate so your call looks something like: > > > > $command = 'api originate user at domain &bridge(user at domain)'; > > > > As for programs able to do that from a Microsoft Product, That I am not sure of. > > > > Jonathan Pitcher > > > > > > > > On 12/16/09 9:59 AM, "John Platts" wrote: > > > > > > > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > > > Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? > > > > _________________________________________________________________ > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From freeswitch at aastral.net Wed Dec 16 08:24:17 2009 From: freeswitch at aastral.net (Bill W) Date: Wed, 16 Dec 2009 11:24:17 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> Message-ID: <4B2909B1.2030103@aastral.net> That's fantastic! FreeSWITCH ROCKS! I'll update the wiki. Thanks, Bill Brian West wrote: > use "apply-proxy-acl" on the sofia profile. > > /b > > On Dec 15, 2009, at 10:58 PM, Bill W wrote: > >> However, having the proxy in the path effectively negates using IP >> based >> ACLS. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 16 09:09:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 09:09:31 -0800 Subject: [Freeswitch-users] Basic Question on the Internal Profile In-Reply-To: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> References: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> Message-ID: <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> On Wed, Dec 16, 2009 at 4:37 AM, Costa Zikalala wrote: > Hi All > > I understand that to connect to a SIP Provider you have to (amongst other > things) define a Gateway on the External Profile. > But some gateways may be defined on the Internal Profile. What kind of > gateways would these be and what would be their purpose as most gateways are > External? > > Hope my question makes a bit of sense. > > This is a good question. First, remember that the profile names "internal" and "external" are just labels that try to give you a basic idea of what you might use them for. The external profile is generally used for outbound registrations, particularly when the FS server is behind NAT. If your FS server isn't behind NAT then you may want to use the internal profile for outbound registrations, although we recommend that your outbound registrations still be in a profile different than your internal profile where local phones are registering. In these cases it's fine just to remove external.xml, make a copy of internal.xml to something like external-nonat.xml, and then edit it so that it uses ports 5080 and 5081. For the record here's what I did for a user just last week: Old: New: Feel free to tinker and remember that if you break something you can just wipe your conf directory and just run "make samples" again and you'll have a brand new set of default configs. Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/3c0e7d09/attachment.html From oscav at hotmail.fr Wed Dec 16 09:57:16 2009 From: oscav at hotmail.fr (Oscav) Date: Wed, 16 Dec 2009 09:57:16 -0800 (PST) Subject: [Freeswitch-users] How to set the Session Name on a SDP? Message-ID: <26815554.post@talk.nabble.com> Hi, Is it possible to set (rewrite) the Session Name in the SDP of a 183 progress sent to inbound ? Many thanks -- View this message in context: http://old.nabble.com/How-to-set-the-Session-Name-on-a-SDP--tp26815554p26815554.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ranjtech at gmail.com Wed Dec 16 10:04:54 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 13:04:54 -0500 Subject: [Freeswitch-users] build errors :( Message-ID: <01c901ca7e7a$47a22900$d6e67b00$@com> Hello All, I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: creating freeswitch cc1: warnings being treated as errors libs/esl/fs_cli.c: In function ?complete?: libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? make[2]: *** [fs_cli-fs_cli.o] Error 1 Any ideas? Thanks RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/ffee79de/attachment.html From brian at freeswitch.org Wed Dec 16 10:13:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 12:13:45 -0600 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01c901ca7e7a$47a22900$d6e67b00$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> Message-ID: <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> What SVN rev? /b On Dec 16, 2009, at 12:04 PM, RR wrote: > Hello All, > > I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: > > creating freeswitch > cc1: warnings being treated as errors > libs/esl/fs_cli.c: In function ?complete?: > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? > make[2]: *** [fs_cli-fs_cli.o] Error 1 > > Any ideas? > > Thanks > RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/30b7491e/attachment-0001.html From bruce.mcalister at blueface.ie Wed Dec 16 10:22:51 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 16 Dec 2009 18:22:51 +0000 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> Message-ID: <4B29257B.10705@blueface.ie> Hi Michael, Michael Jerris wrote: > I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives > Is support for Solaris and/or OpenSolaris x86 planned as well? Thanks Bruce From ranjtech at gmail.com Wed Dec 16 10:27:03 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 13:27:03 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> Message-ID: <01db01ca7e7d$5f14a240$1d3de6c0$@com> Lost that screen that showed me what rev was downloaded but whatever you get after doing a ?svn up? and ?make current?. I had done a rm ?rf in the /usr/src/freeswitch directory and then did svn up. Should I have done svn co instead? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 16, 2009 1:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] build errors :( What SVN rev? /b On Dec 16, 2009, at 12:04 PM, RR wrote: Hello All, I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: creating freeswitch cc1: warnings being treated as errors libs/esl/fs_cli.c: In function ?complete?: libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? make[2]: *** [fs_cli-fs_cli.o] Error 1 Any ideas? Thanks RR __________ Information from ESET NOD32 Antivirus, version of virus signature database 4694 (20091216) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/59d2c37a/attachment.html From mrene_lists at avgs.ca Wed Dec 16 10:34:56 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 16 Dec 2009 13:34:56 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01c901ca7e7a$47a22900$d6e67b00$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> Message-ID: I fixed that already, update to the latest trunk. 32/64 bit types mismatch. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 16-Dec-09, at 1:04 PM, RR wrote: > Hello All, > > I know you will probably ask me to check out a fresh copy from svn > trunk and all, but I assure you I have done that yet I keep getting > these errors on make: > > creating freeswitch > cc1: warnings being treated as errors > libs/esl/fs_cli.c: In function ?complete?: > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long > int?, but argument 4 has type ?int? > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long > int?, but argument 4 has type ?int? > make[2]: *** [fs_cli-fs_cli.o] Error 1 > > Any ideas? > > Thanks > RR > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4694 (20091216) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/98417287/attachment.html From msc at freeswitch.org Wed Dec 16 10:38:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 10:38:14 -0800 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01db01ca7e7d$5f14a240$1d3de6c0$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> Message-ID: <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> On Wed, Dec 16, 2009 at 10:27 AM, RR wrote: > Lost that screen that showed me what rev was downloaded but whatever you > get after doing a ?svn up? and ?make current?. I had done a rm ?rf in the > /usr/src/freeswitch directory and then did svn up. Should I have done svn > co instead? > Why did you nuke the fs src dir? Just curious. In your case since you deleted everything you're probably better off just starting from scratch with a svn co. Once you're installed then all you need to do is "make current" b/c make current includes an "svn up" among other things. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/5743980a/attachment.html From kristian.kielhofner at gmail.com Wed Dec 16 11:13:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:13:38 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? Message-ID: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Hello everyone, Pastebin here: http://pastebin.freeswitch.org/11525 I've got my pjsip profile configured for G722 only: Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Dec 16 11:26:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 13:26:58 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Message-ID: <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> can you do another trace to show the inbound invite too? On Wed, Dec 16, 2009 at 1:13 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > Pastebin here: > > http://pastebin.freeswitch.org/11525 > > I've got my pjsip profile configured for G722 only: > > > > Yet whenever I send calls using that profile it (mysteriously) > indicates support for PCMU in the INVITE. The pastebin includes both > the INVITE and "sofia status profile pjsip" to show that only G722 has > been enabled. Where is PCMU coming from? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/cebcffec/attachment.html From ranjtech at gmail.com Wed Dec 16 11:30:57 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 14:30:57 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> Message-ID: <01f601ca7e86$4c59c730$e50d5590$@com> MC, haha I'm not sure. I think this had happened to me before as well and nuking the fs dir and then an svn up had fixed it. I think I'll just do an svn co and get on with it. Sorry had been following FS when it first started and then for 2 yrs have been busy with a bunch of random stuff but now want to get back into it. BTW, at that time there was no routing/lcr module available for FS and someone, can't remember who had written one to plug into FS but it wasn't open sourced. Wondering if there is one now to use FS more as a class 5 switch than a PBX for more like a carrier peering and minutes wholesale kind of a business?? Sorry if this is a dumb question :( From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 16, 2009 1:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] build errors :( On Wed, Dec 16, 2009 at 10:27 AM, RR wrote: Lost that screen that showed me what rev was downloaded but whatever you get after doing a "svn up" and "make current". I had done a rm -rf in the /usr/src/freeswitch directory and then did svn up. Should I have done svn co instead? Why did you nuke the fs src dir? Just curious. In your case since you deleted everything you're probably better off just starting from scratch with a svn co. Once you're installed then all you need to do is "make current" b/c make current includes an "svn up" among other things. -MC __________ Information from ESET NOD32 Antivirus, version of virus signature database 4694 (20091216) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/d597050a/attachment.html From msc at freeswitch.org Wed Dec 16 11:31:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 11:31:30 -0800 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Message-ID: <87f2f3b90912161131w69cc5daga3ddab8a33b0654a@mail.gmail.com> Try setting "absolute_codec_string" in the dialplan prior to the bridge: Let us know if that does the trick. -MC On Wed, Dec 16, 2009 at 11:13 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > Pastebin here: > > http://pastebin.freeswitch.org/11525 > > I've got my pjsip profile configured for G722 only: > > > > Yet whenever I send calls using that profile it (mysteriously) > indicates support for PCMU in the INVITE. The pastebin includes both > the INVITE and "sofia status profile pjsip" to show that only G722 has > been enabled. Where is PCMU coming from? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/0b7b2910/attachment.html From mgg at giagnocavo.net Wed Dec 16 11:32:40 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 16 Dec 2009 14:32:40 -0500 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <26807322.post@talk.nabble.com> References: <26807322.post@talk.nabble.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> We switched to Windows for production after 1.0.4. We've run into no stability issues with it. The highest we go is only 100 sessions/sec. We're also use media bypass and mod_managed. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 Sent: Wednesday, December 16, 2009 12:39 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] [Windows] Stable enough for production use? Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Dec 16 11:36:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 11:36:57 -0800 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01f601ca7e86$4c59c730$e50d5590$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: <87f2f3b90912161136i2d5bfafeja6b43be1cef8aeab@mail.gmail.com> On Wed, Dec 16, 2009 at 11:30 AM, RR wrote: > MC, haha I?m not sure. I think this had happened to me before as well and > nuking the fs dir and then an svn up had fixed it. > > I think I?ll just do an svn co and get on with it. Sorry had been following > FS when it first started and then for 2 yrs have been busy with a bunch of > random stuff but now want to get back into it. > > > > BTW, at that time there was no routing/lcr module available for FS and > someone, can?t remember who had written one to plug into FS but it wasn?t > open sourced. Wondering if there is one now to use FS more as a class 5 > switch than a PBX for more like a carrier peering and minutes wholesale kind > of a business?? Sorry if this is a dumb question :( > You've got mod_easyroute to handle incoming call routing and mod_lcr for outbound routing. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/daa04363/attachment-0001.html From kristian.kielhofner at gmail.com Wed Dec 16 11:41:00 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:41:00 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> Message-ID: <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> Sure... The call comes up as PCMU: INVITE sip:5888 at 10.70.0.99 SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:9413122830 at smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: , To: "5888" Via: SIP/2.0/UDP 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 Content-Length: 206 Content-Type: application/sdp Contact: Max-Forwards: 70 User-Agent: Avaya CM/R015x.02.0.947.3 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: timer,replaces,join,histinfo,100rel Alert-Info: ;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:9413122830 at smh.sip.local P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: ;index=1,"5888" ;index=1.1 v=0 o=- 1 1 IN IP4 10.70.0.69 s=- c=IN IP4 10.70.0.22 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 We don't support G729 so this call comes up as PCMU when we answer and then that codec is first in the codec list... On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale wrote: > can you do another trace to show the inbound invite too? > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From rupa at rupa.com Wed Dec 16 11:42:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 16 Dec 2009 13:42:51 -0600 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01f601ca7e86$4c59c730$e50d5590$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: I think you are thinking of SWK. He has checked in mod_easyroute for easy routing of DIDs. [intra]lanman and I have mod_lcr for (umm) lcr routing. SWK still has a (probably) more capable lcr - though not open source. On Wed, Dec 16, 2009 at 1:30 PM, RR wrote: > BTW, at that time there was no routing/lcr module available for FS and > someone, can?t remember who had written one to plug into FS but it wasn?t > open sourced. Wondering if there is one now to use FS more as a class 5 > switch than a PBX for more like a carrier peering and minutes wholesale kind > of a business?? Sorry if this is a dumb question :( -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/c60f2bfc/attachment.html From anthony.minessale at gmail.com Wed Dec 16 11:48:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 13:48:20 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> Message-ID: <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding. if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature. On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Sure... The call comes up as PCMU: > > INVITE sip:5888 at 10.70.0.99 SIP/2.0 > Call-ID: 80ea31a017f6de1d53e4a9c52f00 > CSeq: 1 INVITE > From: sip:9413122830 at smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 > Record-Route: , > To: "5888" > > Via: SIP/2.0/UDP > 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP > > 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 > Content-Length: 206 > Content-Type: application/sdp > Contact: > ;transport=tcp> > Max-Forwards: 70 > User-Agent: Avaya CM/R015x.02.0.947.3 > Allow: > INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH > Supported: timer,replaces,join,histinfo,100rel > Alert-Info: > >;avaya-cm-alert-type=external > Min-SE: 1200 > Session-Expires: 1200;refresher=uac > P-Asserted-Identity: sip:9413122830 at smh.sip.local > P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" > History-Info: > >;index=1,"5888" > >;index=1.1 > > v=0 > o=- 1 1 IN IP4 10.70.0.69 > s=- > c=IN IP4 10.70.0.22 > b=AS:64 > t=0 0 > m=audio 2176 RTP/AVP 18 0 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > > We don't support G729 so this call comes up as PCMU when we answer > and then that codec is first in the codec list... > > On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale > wrote: > > can you do another trace to show the inbound invite too? > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8ceb337d/attachment.html From doddlephone at gmail.com Wed Dec 16 11:42:01 2009 From: doddlephone at gmail.com (Doddle WebPhone) Date: Wed, 16 Dec 2009 17:42:01 -0200 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: Message-ID: For an another click2call "flavor/approach", you might use http://www.doddlephone.com to call FS Sergio On Wed, Dec 16, 2009 at 2:24 PM, John Platts wrote: > > You've made my day. > > ________________________________ > > From: jpitcher at nuvio.com > > To: freeswitch-users at lists.freeswitch.org > > Date: Wed, 16 Dec 2009 08:11:05 -0800 > > Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial > > > > > > > > > > > > > > > > > > John, > > > > > > > > To do a click to call in FS you need to have some app that connects to > the ESL or Event Socket Layer and runs one of the calls diagramed here ... > > > > > > > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > > > > > > > For use with the ESL just prepend api in front of the originate so your > call looks something like: > > > > > > > > $command = 'api originate user at domain &bridge(user at domain)'; > > > > > > > > As for programs able to do that from a Microsoft Product, That I am not > sure of. > > > > > > > > Jonathan Pitcher > > > > > > > > > > > > > > > > On 12/16/09 9:59 AM, "John Platts" wrote: > > > > > > > > > > > > > > > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > > > > > > > Do you have any recommendations on programs capable of click-to-call or > click-to-dial from Microsoft Outlook or Microsoft Excel? > > > > > > > > _________________________________________________________________ > > > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > > > > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________ > Hotmail: Trusted email with powerful SPAM protection. > http://clk.atdmt.com/GBL/go/177141665/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/1bfad4f3/attachment.html From kristian.kielhofner at gmail.com Wed Dec 16 11:59:17 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:59:17 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> Message-ID: <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> Anthony, As always, thanks. I thought that might be it but I wanted to make sure. Thanks again! On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale wrote: > yah so the codec chosen by the inbound leg is always offered in the outbound > sdp to try and prevent transcoding. > if you set {absolute_codec_string=G722} in the bridge string you will bypass > this feature. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From ranjtech at gmail.com Wed Dec 16 12:00:44 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 15:00:44 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: <020e01ca7e8a$756bb4e0$60431ea0$@com> I think you are thinking of SWK. He has checked in mod_easyroute for easy routing of DIDs. [intra]lanman and I have mod_lcr for (umm) lcr routing. SWK still has a (probably) more capable lcr - though not open source. On Wed, Dec 16, 2009 at 1:30 PM, RR wrote: BTW, at that time there was no routing/lcr module available for FS and someone, can't remember who had written one to plug into FS but it wasn't open sourced. Wondering if there is one now to use FS more as a class 5 switch than a PBX for more like a carrier peering and minutes wholesale kind of a business?? Sorry if this is a dumb question :( -- -Rupa Thanks Rupa and MC! Will check out these modules. I'm hoping/assuming there's some documentation available for these? If not, then would you be able to share some sample configs that work? Thanks \RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/5a3215f6/attachment-0001.html From anthony.minessale at gmail.com Wed Dec 16 12:16:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 14:16:42 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> Message-ID: <191c3a030912161216o70053862y9c6d3c33be43efe8@mail.gmail.com> np On Wed, Dec 16, 2009 at 1:59 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Anthony, > > As always, thanks. I thought that might be it but I wanted to make sure. > > Thanks again! > > On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale > wrote: > > yah so the codec chosen by the inbound leg is always offered in the > outbound > > sdp to try and prevent transcoding. > > if you set {absolute_codec_string=G722} in the bridge string you will > bypass > > this feature. > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/34ba02af/attachment.html From costa.zikalala at gmail.com Wed Dec 16 12:17:17 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Wed, 16 Dec 2009 22:17:17 +0200 Subject: [Freeswitch-users] Basic Question on the Internal Profile In-Reply-To: <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> References: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> Message-ID: <59daa2cd0912161217r352942aak94ee04f770b1ced5@mail.gmail.com> Thanks for the detailed response Michael, it certainly helped a lot. 2009/12/16 Michael Collins > > > On Wed, Dec 16, 2009 at 4:37 AM, Costa Zikalala wrote: > >> Hi All >> >> I understand that to connect to a SIP Provider you have to (amongst other >> things) define a Gateway on the External Profile. >> But some gateways may be defined on the Internal Profile. What kind of >> gateways would these be and what would be their purpose as most gateways are >> External? >> >> Hope my question makes a bit of sense. >> >> > This is a good question. First, remember that the profile names "internal" > and "external" are just labels that try to give you a basic idea of what you > might use them for. The external profile is generally used for outbound > registrations, particularly when the FS server is behind NAT. If your FS > server isn't behind NAT then you may want to use the internal profile for > outbound registrations, although we recommend that your outbound > registrations still be in a profile different than your internal profile > where local phones are registering. In these cases it's fine just to remove > external.xml, make a copy of internal.xml to something like > external-nonat.xml, and then edit it so that it uses ports 5080 and 5081. > For the record here's what I did for a user just last week: > > Old: > > > value="$${internal_ssl_enable}"/> > > value="transport=tls"/> > > value="$${internal_tls_port}"/> > > value="$${internal_ssl_dir}"/> > > > > New: > > > value="$${external_ssl_enable}"/> > > value="$${external_tls_port}"/> > > value="$${external_ssl_dir}"/> > > > Feel free to tinker and remember that if you break something you can just > wipe your conf directory and just run "make samples" again and you'll have a > brand new set of default configs. > > Enjoy! > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/99220676/attachment.html From brian at proximosystems.com Wed Dec 16 13:49:09 2009 From: brian at proximosystems.com (Brian) Date: Wed, 16 Dec 2009 16:49:09 -0500 Subject: [Freeswitch-users] mod_conference scalability Message-ID: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8cd81a6d/attachment.html From jerry.richards at teotech.com Wed Dec 16 14:23:11 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 16 Dec 2009 14:23:11 -0800 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 Message-ID: I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" reply. Below is the console loglevel 7 output. It says the destination is out-of-order. I'm not sure what this means. Any help is appreciated. 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0] 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by acl "domains". Falling back to Digest auth. 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0] 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by acl "domains". Falling back to Digest auth. 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel sofia/internal/5381 at 192.168.72.141:5060 [e58e763f-7688-4600-aa70-481bbc359f58] 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: v=0 o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4 192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000/1 a=ptime:20 a=ptime:20 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> CS_INIT 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/5381 at 192.168.72.141:5060) State INIT 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> CS_ROUTING 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_ROUTING 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing Anonymous->93491028 in context default Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->unloop] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->tod_example] continue=true Dialplan: day of week[4] =~ 2-6 (PASS) Dialplan: hour[14] =~ 9-18 (PASS) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(open=true) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->holiday_example] continue=true Dialplan: month[12] =~ 1 (FAIL) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->Mediant1000] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [Mediant1000] destination_number(93491028) =~ /^8(\d+)$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->SangomaPRI] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [SangomaPRI] destination_number(93491028) =~ /^9(\d+)$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(effective_caller_id_number=425740${caller_id_number}) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action bridge(openzap/smg_prid/a/3491028 at g1) 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> CS_EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(open=true) 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(effective_caller_id_number=4257405381) 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 sofia/internal/5381 at 192.168.72.141:5060 SET [effective_caller_id_number]=[4257405381] EXECUTE sofia/internal/5381 at 192.168.72.141:5060 bridge(openzap/smg_prid/a/3491028 at g1) 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/5381 at 192.168.72.141:5060 [KILL] 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_HANGUP 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to INVITE with: 502 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> CS_REPORTING 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_REPORTING 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to sleep 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> CS_DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session 1 (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external entities 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session 1 (sofia/internal/5381 at 192.168.72.141:5060) Ended 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close Channel sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep Best Regards, Jerry From brian at freeswitch.org Wed Dec 16 14:27:52 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 16:27:52 -0600 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: References: Message-ID: Need siptrace with this type "sofia profile xxxx siptrace on" replace xxxx with your profile. /b On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote: > I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal > phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" > reply. Below is the console loglevel 7 output. It says the destination is > out-of-order. I'm not sure what this means. Any help is appreciated. > > 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0] > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by > acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0] > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by > acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/5381 at 192.168.72.141:5060 > [e58e763f-7688-4600-aa70-481bbc359f58] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel > sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: > v=0 > o=TC 1100638826 1100638826 IN IP4 192.168.72.32 > s=session > c=IN IP4 192.168.72.32 > t=0 0 > m=audio 1760 RTP/AVP 0 18 4 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000/1 > a=ptime:20 > a=ptime:20 > > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING > 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing > Anonymous->93491028 in context default > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(open=true) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->Mediant1000] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [Mediant1000] > destination_number(93491028) =~ /^8(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->SangomaPRI] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [SangomaPRI] > destination_number(93491028) =~ /^9(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(effective_caller_id_number=425740${caller_id_number}) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(open=true) > 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > set(effective_caller_id_number=4257405381) > 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET > [effective_caller_id_number]=[4257405381] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string > 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot create > outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 Originate > Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. > Cause: DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup > sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [KILL] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to INVITE > with: 502 > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to sleep > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session 1 > (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external > entities > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session 1 > (sofia/internal/5381 at 192.168.72.141:5060) Ended > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close Channel > sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From moises.silva at gmail.com Wed Dec 16 15:26:49 2009 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 16 Dec 2009 18:26:49 -0500 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> Message-ID: I've been using FreeSWITCH on Windows lately and seems to work pretty well. Sangoma has been testing more and more lately the Windows drivers with FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 though, always using trunk, if you are going to be using PSTN lines (and therefore requiring openzap) I think it would be a good idea for you to use trunk and latest wanpipe drivers. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Wed, Dec 16, 2009 at 2:32 PM, Michael Giagnocavo wrote: > We switched to Windows for production after 1.0.4. We've run into no > stability issues with it. The highest we go is only 100 sessions/sec. We're > also use media bypass and mod_managed. > > -Michael > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 > Sent: Wednesday, December 16, 2009 12:39 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] [Windows] Stable enough for production use? > > > Hello > > Since Freeswitch is also available for Windows (and Mac, but I don't > anything about Macintosh), I'd like some feedback from users who routinely > run Freeswitch on that OS. > > Is it stable enough to be used in production to handle a single analog line > (ie. SOHO use), or should I warn customers that they really should buy a > dedicated Linux box to run FS? > > Thank you. > -- > View this message in context: > http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/fe91295b/attachment.html From djbinter at yahoo.com Wed Dec 16 15:29:10 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 16 Dec 2009 15:29:10 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite Message-ID: <607753.94827.qm@web37503.mail.mud.yahoo.com> We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/e9398029/attachment.html From anthony.minessale at gmail.com Wed Dec 16 15:42:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 17:42:48 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <607753.94827.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> Message-ID: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and here is the > issue: > > > > Call is initially setup fine and they send a first re-invite with media > 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first > re-invite fine > > > > They then send a second re-invite with their media IP to cut through media > and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it > is strictly for the purpose of ?keep alive? per RFC 4028. This third > re-invite has the exact same media IP and UDP pot information as the second > re-invite does. The problem is FS does not respond to this third re-invite > AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a response from FS. > > > One more thing, we did not see the third re-invite in sofia siptrace, but > we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > > Thank you very much. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/067dd91c/attachment-0001.html From msc at freeswitch.org Wed Dec 16 16:15:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 16:15:38 -0800 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> And we shouldn't be using 1.0.4 anyway, should we? ;) -MC On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva wrote: > I've been using FreeSWITCH on Windows lately and seems to work pretty well. > Sangoma has been testing more and more lately the Windows drivers with > FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 > though, always using trunk, if you are going to be using PSTN lines (and > therefore requiring openzap) I think it would be a good idea for you to use > trunk and latest wanpipe drivers. > > -- > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8ee37d7b/attachment.html From msc at freeswitch.org Wed Dec 16 16:20:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 16:20:12 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00aa01ca7e99$9901f9a0$cb05ece0$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: <87f2f3b90912161620p627d676i85a53c3771d87a91@mail.gmail.com> how are the listeners connecting? On Wed, Dec 16, 2009 at 1:49 PM, Brian wrote: > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/4afd8965/attachment.html From steveu at coppice.org Wed Dec 16 16:48:21 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Dec 2009 08:48:21 +0800 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <4B29257B.10705@blueface.ie> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> <4B29257B.10705@blueface.ie> Message-ID: <4B297FD5.5060909@coppice.org> On 12/17/2009 02:22 AM, Bruce McAlister wrote: > Hi Michael, > > Michael Jerris wrote: > >> I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives >> >> > Is support for Solaris and/or OpenSolaris x86 planned as well? > > Thanks > Bruce > Support for the less popular platforms will probably depend mostly on the availability of machines for building and testing. The code is unlikely to ever be as well optimised for speed on the less popular platforms, but basic support should be largely an issue of access for development and support. Steve From djbinter at yahoo.com Wed Dec 16 17:00:11 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 16 Dec 2009 17:00:11 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> Message-ID: <800257.53977.qm@web37503.mail.mud.yahoo.com> Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: We >have a customer that we are sending calls to off the FS and here is the issue: > >Call >is initially setup fine and they send a first re-invite with media 0.0.0.0 to >place the caller on hold. FS sends a 200 ok to this first re-invite fine > >They >then send a second re-invite with their media IP to cut through media and the >FS sends a 200 OK to this fine. At this point the call is fine > >30 >minutes later they send a third re-invite because according to them it is >strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has >the exact same media IP and UDP pot information as the second re-invite does. >The problem is FS does not respond to this third re-invite AT ALL. It doesn?t >send a 100 trying a 200 OK nothing so this causes the call to be dropped as the >other end does not recieve a response from FS. > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > >Thank you very much. > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/e49d756e/attachment.html From senakahks at gmail.com Wed Dec 16 18:44:31 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Wed, 16 Dec 2009 18:44:31 -0800 (PST) Subject: [Freeswitch-users] Billing solutions information In-Reply-To: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> References: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> Message-ID: <1261017871151-4179366.post@n2.nabble.com> Hi, Seems nobody is interested to talk about this topic. I found nibblebill is great. But doesn't hangup the call when balance goes to zero. The other problem It allows user to call without checking the balance of the cash database. Is this natural?. If this works fine we can easily integrate with a payment gateway like 2checkout. Thank you Lon Baker wrote: > > Hey everyone, > > I am researching billing solutions for Freeswitch and want to consolidate > the information with what others have found, then add it to the Wiki. > > There are seems to be a number of billing solutions by commercial > providers, > claiming they can integrate with Freeswitch, but nothing concrete > explaining > how far they go. > > Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, > etc? > > Mod_nibblebill handles the basics of updating a database table. > > The A2Billing teams says they are planning on adding support for > Freeswitch > in a few months. > > ASTPP.org says they support Freeswitch, but the site hasn't been updated > since 2008. > > If you know about any solutions, links to solutions or any information can > you send it to me? I will organize it and add it to the wiki. > > Thanks! > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Billing-solutions-information-tp4166151p4179366.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mgg at giagnocavo.net Wed Dec 16 20:25:19 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 16 Dec 2009 23:25:19 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <800257.53977.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <800257.53977.qm@web37503.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C42866F@mse17be1.mse17.exchange.ms> FWIW, we?ve seen the same thing intermittently, haven?t had time/been able to get a solid repro to capture debug information. Call ID and tags are all matching. After the re-invite fails and the remote end sends a BYE, FS does indeed respond to the re-invite. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of DJB Sent: Wednesday, December 16, 2009 6:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP Re-invite Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/ce3e621f/attachment-0001.html From yehavi.bourvine at gmail.com Wed Dec 16 21:16:38 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 07:16:38 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? Message-ID: Hello, I am trying to debug a TLS handshake error between FreeSwitch and some ATA. When setting the loglevel to 9 I get only a message that TLS handshake failed. Is there some other debug command to show what happens during the TLS handshake process? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/c53fbb66/attachment.html From yehavi.bourvine at gmail.com Wed Dec 16 21:39:49 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 07:39:49 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: An interim update: - *Audiocodes*: No success yet. I am working with the manufacturer to debug it. - *VegaStream:* Got the necessary license, configured TLS but it doesn't work. I am working with the local representatives on it. Regards, __Yehavi: 2009/12/10 Brian West > I have confirmed it works with Polycom, Snom and a few others .... > polycom is the hardest to set due to having to put the ca cert into > the phone... but other than that its good. > > /b > > On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > > > An intermediate report: > > > > Audiocodes: TLS works only on outgoing requests, incoming ones are > > ignored. I am waiting for Audiocodes' help in order to debug it. > > SRTP: worked when no TLS is active. When TLS is active the call is > > disconnected when the remote party answers. Still debugging it. > > > > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > > how to enable TLS from the WEB interface. > > > > Regards, __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/708106d7/attachment.html From juanbackson at gmail.com Wed Dec 16 21:53:36 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 17 Dec 2009 13:53:36 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Message-ID: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Hi I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. Is there any other ways to check for zombie channels? jb On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: > Why not just set rtp-timeout-sec on the sofia profile and it'll do > that for you. > > Unless something else is going on. > > /b > > On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > > > Hi, > > > > I am having problem with around 1 % of the channels always get > > zombilized. > > > > What I want to do is to have a background thread that regularly > > check all the channels that have been in existance for like > 1 hr, > > and then check to see if there is any RTP coming in and going out. > > If there is no RTP, then I just hangup that channel. Does anyone > > know if there is anyway to do that in a freeswitch module? Which > > API can I use to accomplish this purpose? Alternatively, is there > > anyway to configure freeswitch so that it will hangup the calls > > where there is no media in and out for so many seconds? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7cdefe03/attachment.html From mcampbellsmith at gmail.com Wed Dec 16 22:39:56 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 17 Dec 2009 17:39:56 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> Thanks Yehavi... I posted a question on the Cisco Forum and am waiting a response from 'engineering' to tell us if they plan to implement standard SRTP support in the Linksys ATA's. TLS is working fine. On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine wrote: > An interim?update: > > > Audiocodes: No success yet.?I am working with the manufacturer to debug it. > VegaStream: Got the necessary license, configured TLS but it doesn't work. I > am working with the local representatives on it. > > ????????????????????????????? Regards, __Yehavi: > > 2009/12/10 Brian West >> >> I have confirmed it works with Polycom, Snom and a few others .... >> polycom is the hardest to set due to having to put the ca cert into >> the phone... but other than that its good. >> >> /b >> >> On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: >> >> > An intermediate report: >> > >> > Audiocodes: TLS works only on outgoing requests, incoming ones are >> > ignored. I am waiting for Audiocodes' help in order to debug it. >> > SRTP: worked when no TLS is active. When TLS is active the call is >> > disconnected when the remote party answers. Still debugging it. >> > >> > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions >> > how to enable TLS from the WEB interface. >> > >> > ? ? ? ? ? ? ? ? ? ? ? ? ?Regards, __Yehavi: >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From yehavi.bourvine at gmail.com Wed Dec 16 23:29:43 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 09:29:43 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: 2009/11/29 Ujjval Karihaloo > Polycom Firmware matrix (Look at the polycom website) does not allow > firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones?So > first confirm you are on a supported firmware release? > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi > Bourvine > *Sent:* Sunday, November 29, 2009 8:48 AM > *To:* freeswitch-users > *Subject:* [Freeswitch-users] Polycom 501 conferencing with FreeSwitch > > > > Hello, > > > > I am trying to set a Polycom 501 phone to do conferencing via the > conference room on Freeswitch rather than on the phone (as on the phone it > is limited to 3 participants only). Anyone had success with it? > > > > I have on the Freeswitch an extension named Conf.* which activates the > conference application (it works with other brands). On the Polycom I tried > to define > > voIpProt.SIP.*conference*.address=sip:Conf0000 at freeswitch-server. The > phone continues to create the conference locally and add the above Conf0000 > to it, without REFERing the parties to it. The first phone which called is > left on hold... > > > > Anyone managed to make this feature work? We use firmware 3.1.3. > > > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/a57eba73/attachment.html From talk2ram at gmail.com Thu Dec 17 02:29:10 2009 From: talk2ram at gmail.com (ram) Date: Thu, 17 Dec 2009 02:29:10 -0800 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: Hi Look at Contrib of source http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ some pre-paid examples Ram On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi wrote: > Dear Sir, > > I have successfully installed freeSWITCH and it works fine in passthrough > mode. I installed nibblebill and it deduct money from the accounts database > and it works fine. but I have two problems. > > 1. Calls can be initiated even though there is a minus value in accounts > database > > 2. Calls doesn't hangup when it goes to minus values. > > Any answers are greatly appreciated. > > This is my dialplan: > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/9e1ae678/attachment-0001.html From yivzhenko at mksat.net Thu Dec 17 03:05:56 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 17 Dec 2009 13:05:56 +0200 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 Message-ID: <200912171305.57498.yivzhenko@mksat.net> Hallo All! I need information about setup BLF on GXP2010/2020 phones with Freeswitch. I search in Freeswitch Wiki and maillist archives but find no usable information. From oscav at hotmail.fr Thu Dec 17 03:21:21 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 17 Dec 2009 03:21:21 -0800 (PST) Subject: [Freeswitch-users] How to set the Session Name on a SDP? In-Reply-To: <26815554.post@talk.nabble.com> References: <26815554.post@talk.nabble.com> Message-ID: <26826579.post@talk.nabble.com> I just found that this is related to the username of the profile. It needs to be set as parameter. Oscav wrote: > > Hi, > > Is it possible to set (rewrite) the Session Name in the SDP of a 183 > progress sent to inbound ? > > Many thanks > -- View this message in context: http://old.nabble.com/How-to-set-the-Session-Name-on-a-SDP--tp26815554p26826579.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 17 03:41:54 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 06:41:54 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> Message-ID: <4B2A1902.2050008@aastral.net> Okay, I added: to my sofia profile and restarted sofia, and still no joy. I'm on FreeSWITCH Version 1.0.trunk (15764) I've got in the directory, but I'm still being rejected by the acl: 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.103.12/32 Here's what I believe is the appropriate snippet of the debug output: http://pastebin.freeswitch.org/11531 Thoughts? Thanks, Bill Brian West wrote: > use "apply-proxy-acl" on the sofia profile. > > /b > > On Dec 15, 2009, at 10:58 PM, Bill W wrote: > >> However, having the proxy in the path effectively negates using IP >> based >> ACLS. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From senakahks at gmail.com Thu Dec 17 03:53:03 2009 From: senakahks at gmail.com (Senaka Amarakeerthi) Date: Thu, 17 Dec 2009 20:53:03 +0900 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: Dear Ram, Thank you for the reply. To work with your code I hope that Mod cdr should be there. But wiki says that its not functional. What should I do. Thanks Senaka On Thu, Dec 17, 2009 at 7:29 PM, ram wrote: > Hi > > Look at Contrib of source > > http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ > > some pre-paid examples > > Ram > > On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi > wrote: >> >> Dear Sir, >> >> I have successfully installed freeSWITCH and it works fine in passthrough >> mode. I installed nibblebill and it deduct money from the accounts >> database >> and it works fine. but I have two problems. >> >> 1. Calls can be initiated even though there is a minus value in accounts >> database >> >> 2. Calls doesn't hangup when it goes to minus values. >> >> Any answers are greatly appreciated. >> >> This is my dialplan: >> >> >> >> >> ? >> ? ? >> ? ? >> ? >> >> ? >> >> >> >> >> >> > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> >> >> >> >> >> >> This is the configuration file; >> >> >> ? >> ? ? >> >> ? ? >> >> >> >> >> ? ? >> >> >> ? ? >> >> >> ? ? >> >> >> >> ? ? >> >> >> ? ? >> >> >> >> ? ? >> >> >> >> ? ? >> >> >> >> ? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From saeedahmad1981 at gmail.com Thu Dec 17 03:54:45 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 17 Dec 2009 12:54:45 +0100 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> Message-ID: with the scenario below can we get the better performance: We create one profile for incoming call listening on 5060 as profile1 we create two profile for outgoing calls as profile2 on 5050 and profile3 on 5051 now we are receiving all calls on profile1:5060, but while bridging them to vendors we divide them, half to profile2:5050 and half to profile3:5051, something like: Will it make any difference? Thanks On Sun, Dec 13, 2009 at 11:37 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Sep processes does better than sep profiles. We need to push the sofia > devs to work on a better concurrancy scheme but they are too busy with other > nokia duties these days so were stuck with what we got for now. About > 400cps on a good day > > On Dec 13, 2009 4:05 PM, "Jay Binks" wrote: > > I'm interested in what the upper limit would be, when expecting a > performance improvement with sofia profiles. > > For example let's say I were to direct connect to customers ( layer 2 ) > with a .1q trunk coming in to fs and a Sofia profile for each customer. Am > I going to hit a bottleneck at 20,50,100,500 ??? > > Guess it's hardware limited , but any thoughts ? > > J > > On 14/12/2009, at 4:36, Anthony Minessale > wrote: > Here is my standa... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/37563e6f/attachment.html From codecomplete at free.fr Thu Dec 17 04:54:49 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 04:54:49 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <72220.45962.qm@web111310.mail.gq1.yahoo.com> References: <26808383.post@talk.nabble.com> <72220.45962.qm@web111310.mail.gq1.yahoo.com> Message-ID: <26827581.post@talk.nabble.com> I don't have access to a remote computer from which I could log on and run nmap. I'll see if I can get a shell access somewhere. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From foxb at abv.bg Thu Dec 17 05:14:41 2009 From: foxb at abv.bg (Hristo Benev) Date: Thu, 17 Dec 2009 15:14:41 +0200 (EET) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Message-ID: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Just for your information there is a version of nmap for windows. So you can do the test from your desktop. >-------- ?????????? ????? -------- >??: Fred-145 >???????: Re: [Freeswitch-users] Scanning my firewall for open UDP ports? >??: freeswitch-users at lists.freeswitch.org >????????? ??: ?????????, 2009, ???????? 17 14:54:49 EET > >I don't have access to a remote computer from which I could log on and run >nmap. > >I'll see if I can get a shell access somewhere. Thank you. >-- >View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html >Sent from the Freeswitch-users mailing list archive at Nabble.com. > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > From neilp at cs.stanford.edu Thu Dec 17 05:34:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 17 Dec 2009 19:04:58 +0530 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: Hi Mike, This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. In case this wasn't apparent I am trying to install FS from trunk. Thanks, Neil On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote: > strange, can someone file a bug on this on jira.freeswitch.org and contact > me off list with ssh info so I can troubleshoot this on your box. > > Thanks > Mike > > On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > > I'm also experiencing this problem, and I have verified I have libogg, > libvorbis, and their dev packages installed. > > I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not > listed in the dependency lib list. Is this related? > > -Neil > > On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > >> looks like ogg devel packages are installed but ogg lib is not? >> >> >> On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: >> >> > FreeSWITCH seems to be unable to read MP3 files, citing that it's an >> > unknown format. Looking through the log, I found this during startup: >> > >> > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error >> > Loading module /usr/local/freeswitch/mod/mod_shout.so >> > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: >> > ogg_sync_wrote** >> > >> > There don't seem to be any compile-time errors, yet I can't seem to >> > eliminate this issue. Any help would be appreciated. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/1cf3832c/attachment-0001.html From brian at freeswitch.org Thu Dec 17 06:50:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:50:35 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Message-ID: <7C2A7FEA-BB01-4176-B64D-776C40565F01@freeswitch.org> We need more info... svn rev, gcore, back trace and what not... please see the reporting bugs link on the wiki. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Dec 16, 2009, at 11:53 PM, Juan Backson wrote: > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb From brian at freeswitch.org Thu Dec 17 06:51:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:51:05 -0600 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2A1902.2050008@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> Message-ID: <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> it needs to be an ACL from acl.conf or a ip/cidr /b On Dec 17, 2009, at 5:41 AM, Bill W wrote: > Okay, I added: to my sofia > profile and restarted sofia, and still no joy. > > I'm on FreeSWITCH Version 1.0.trunk (15764) > I've got in > the directory, but I'm still being rejected by the acl: > > 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 > Rejected by user acl 190.218.103.12/32 > > Here's what I believe is the appropriate snippet of the debug output: > http://pastebin.freeswitch.org/11531 > > Thoughts? > Thanks, > Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/aba9c27b/attachment.html From brian at freeswitch.org Thu Dec 17 06:51:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:51:27 -0600 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: Works on my CentOS 5.4 box just fine... /b On Dec 17, 2009, at 7:34 AM, Neil Patel wrote: > Hi Mike, > > This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. > > In case this wasn't apparent I am trying to install FS from trunk. > > Thanks, > Neil From brian at freeswitch.org Thu Dec 17 07:00:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 09:00:54 -0600 Subject: [Freeswitch-users] How to set the Session Name on a SDP? In-Reply-To: <26826579.post@talk.nabble.com> References: <26815554.post@talk.nabble.com> <26826579.post@talk.nabble.com> Message-ID: <3488E7DE-395B-41A4-A65D-73C0ACC33358@freeswitch.org> Why are you needing to change it? /b On Dec 17, 2009, at 5:21 AM, Oscav wrote: > > I just found that this is related to the username of the profile. It needs to > be set as parameter. From mike at jerris.com Thu Dec 17 07:18:25 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:18:25 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00aa01ca7e99$9901f9a0$cb05ece0$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn?t scale nearly as well as I?d hoped (based on what I?ve read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I?ve set file logging to ?notice? level. My conference profile is configured to suppress several events, hoping that it would improve performance. > > Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: > > Scenario 1: > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/d8361473/attachment.html From mike at jerris.com Thu Dec 17 07:36:44 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:36:44 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <607753.94827.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> Message-ID: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and here is the issue: > > > > Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. > > > > One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ce45758c/attachment-0001.html From mike at jerris.com Thu Dec 17 07:39:03 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:39:03 -0500 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). > > Any idea whether it is possible to program Freeswitch to support this draft? > > Thanks, __Yehavi: > From djbinter at yahoo.com Thu Dec 17 07:48:54 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 07:48:54 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> Message-ID: <206417.45016.qm@web37507.mail.mud.yahoo.com> Anthony, I have pasted the invite sip trace here:? http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: >? >Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >? >They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >? >30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > >Thank you very much. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/636afbe0/attachment.html From mayamatakeshi at gmail.com Thu Dec 17 07:53:54 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 18 Dec 2009 00:53:54 +0900 Subject: [Freeswitch-users] Small delay in registration validity Message-ID: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> It seems to me, in previous revisions of FS, we could successfully call a registered user as soon as his terminal gets 200 OK for REGISTER. But after testing recent revisions, it seems we must wait a little (I wait 1 second) otherwise a call to bridge would end with this: 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] Similar thing is happening when the terminal unregisters: after unregistration an immediate call to bridge sofia/profile/user%domain will succeed. Has anything changed recently in the way registration works that could explain this? br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/fc45cb14/attachment.html From anthony.minessale at gmail.com Thu Dec 17 07:54:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 09:54:21 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> Message-ID: <191c3a030912170754l4589fb94v190820b51c39bada@mail.gmail.com> Is the packet capture running on the FS box itself? On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris wrote: > if you don't see it in sofia siptrace but do see it in tcpdump capture then > something very ugly is going on. Either sofia has hung up completely and is > not listening on that port anymore (can other calls go through?) or the > packet you see in tcpdump is not really going to the right port. Can you > confirm which one? > > Mike > > On Dec 16, 2009, at 6:29 PM, DJB wrote: > > We have a customer that we are sending calls to off the FS and here is the > issue: > > > > Call is initially setup fine and they send a first re-invite with media > 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first > re-invite fine > > > > They then send a second re-invite with their media IP to cut through media > and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it > is strictly for the purpose of ?keep alive? per RFC 4028. This third > re-invite has the exact same media IP and UDP pot information as the second > re-invite does. The problem is FS does not respond to this third re-invite > AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a response from FS. > > > One more thing, we did not see the third re-invite in sofia siptrace, but > we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b91a98b0/attachment.html From anthony.minessale at gmail.com Thu Dec 17 07:57:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 09:57:42 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > ------------------------------ > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wed, December 16, 2009 3:42:48 PM > *Subject:* Re: [Freeswitch-users] SIP Re-invite > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >> We have a customer that we are sending calls to off the FS and here is >> the issue: >> >> >> >> Call is initially setup fine and they send a first re-invite with media >> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first >> re-invite fine >> >> >> >> They then send a second re-invite with their media IP to cut through media >> and the FS sends a 200 OK to this fine. At this point the call is fine >> >> >> >> 30 minutes later they send a third re-invite because according to them it >> is strictly for the purpose of ?keep alive? per RFC 4028. This third >> re-invite has the exact same media IP and UDP pot information as the second >> re-invite does. The problem is FS does not respond to this third re-invite >> AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the >> call to be dropped as the other end does not recieve a response from FS. >> >> >> One more thing, we did not see the third re-invite in sofia siptrace, but >> we do see it in ethereal, which is kind of odds. >> >> >> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >> Thank you very much. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/6a1d4b55/attachment-0001.html From mike at jerris.com Thu Dec 17 07:58:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:58:42 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: <5CB306F6-3868-41CA-B633-F7809D1540B8@jerris.com> if you contact me offlist, or better, join #freeswitch on irc.freenode.net and ping me (MikeJ) Mike On Dec 17, 2009, at 8:34 AM, Neil Patel wrote: > Hi Mike, > > This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. > > In case this wasn't apparent I am trying to install FS from trunk. > > Thanks, > Neil > > On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote: > strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. > > Thanks > Mike > > On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > >> I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. >> >> I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? >> >> -Neil >> >> On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: >> looks like ogg devel packages are installed but ogg lib is not? >> >> >> On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: >> >> > FreeSWITCH seems to be unable to read MP3 files, citing that it's an >> > unknown format. Looking through the log, I found this during startup: >> > >> > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error >> > Loading module /usr/local/freeswitch/mod/mod_shout.so >> > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: >> > ogg_sync_wrote** >> > >> > There don't seem to be any compile-time errors, yet I can't seem to >> > eliminate this issue. Any help would be appreciated. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/5e13de31/attachment.html From mrene_lists at avgs.ca Thu Dec 17 08:00:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 17 Dec 2009 11:00:12 -0500 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Message-ID: <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> Are you doing proxy or bypass meda? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Dec-09, at 12:53 AM, Juan Backson wrote: > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls > with duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb > > On Wed, Dec 16, 2009 at 10:52 PM, Brian West > wrote: > Why not just set rtp-timeout-sec on the sofia profile and it'll do > that for you. > > Unless something else is going on. > > /b > > On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > > > Hi, > > > > I am having problem with around 1 % of the channels always get > > zombilized. > > > > What I want to do is to have a background thread that regularly > > check all the channels that have been in existance for like > 1 hr, > > and then check to see if there is any RTP coming in and going out. > > If there is no RTP, then I just hangup that channel. Does anyone > > know if there is anyway to do that in a freeswitch module? Which > > API can I use to accomplish this purpose? Alternatively, is there > > anyway to configure freeswitch so that it will hangup the calls > > where there is no media in and out for so many seconds? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7ad25dcb/attachment.html From mike at jerris.com Thu Dec 17 08:03:46 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 11:03:46 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/170d35d9/attachment.html From kristian.kielhofner at gmail.com Thu Dec 17 08:04:02 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 11:04:02 -0500 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: Message-ID: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> You could try ssldump: http://www.rtfm.com/ssldump/ On Thu, Dec 17, 2009 at 12:16 AM, Yehavi Bourvine wrote: > Hello, > > ? I am trying to debug a TLS handshake error between FreeSwitch and some > ATA.?When setting the loglevel to 9 I get only a message that TLS handshake > failed. Is there some other debug command to show what happens during the > TLS handshake process? > > ??????????????????????????? Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From yehavi.bourvine at gmail.com Thu Dec 17 08:07:41 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 18:07:41 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... Thanks, __Yehavi: 2009/12/17 Michael Jerris > Its software, anything is possible with enough time and effort. > > Mike > > On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > > > After some discussions with Polycom support it seems that their > conferencing support is based on draft-ietf-sipping-cc-conferencing-03 > (which is not the latest and is not compatible with the latest one). > > > > Any idea whether it is possible to program Freeswitch to support this > draft? > > > > Thanks, __Yehavi: > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7a60083a/attachment-0001.html From brian at freeswitch.org Thu Dec 17 08:08:38 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 10:08:38 -0600 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> Message-ID: <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: > You could try ssldump: > > http://www.rtfm.com/ssldump/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/42dccdc8/attachment.html From djbinter at yahoo.com Thu Dec 17 08:11:34 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 08:11:34 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> Message-ID: <556038.39248.qm@web37502.mail.mud.yahoo.com> The trace that I pasted on the pastebin was from?our analyzer,Tektronix spectra2 that was sitting between FS and customer.? I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here:? http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >>? >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >>? >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >>? >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >pstn:?+19193869900? +19193869900 > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/eea48cca/attachment.html From anthony.minessale at gmail.com Thu Dec 17 08:12:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 10:12:58 -0600 Subject: [Freeswitch-users] Small delay in registration validity In-Reply-To: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> References: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> Message-ID: <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> The sql is sorted into transactions to boost performance so it waits for either 500 statements to execute or 500ms to elapse to accumulate as many sql stmts as possible into the transaction. set sql-in-transactions to false in your profile or make a patch to make the 500ms configurable On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote: > It seems to me, in previous revisions of FS, we could successfully call a > registered user as soon as his terminal gets 200 OK for REGISTER. > But after testing recent revisions, it seems we must wait a little (I wait > 1 second) otherwise a call to bridge would end with this: > > 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create > outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > > Similar thing is happening when the terminal unregisters: after > unregistration an immediate call to bridge sofia/profile/user%domain will > succeed. > > Has anything changed recently in the way registration works that could > explain this? > > br, > takeshi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/18f2f840/attachment.html From dave at 3c.co.uk Thu Dec 17 08:13:30 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 16:13:30 +0000 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <1261066410.6396.73.camel@local.freepabx.com> I'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and > here is the issue: > > > > Call is initially setup fine and they send a first re-invite > with media 0.0.0.0 to place the caller on hold. FS sends a 200 > ok to this first re-invite fine > > > > They then send a second re-invite with their media IP to cut > through media and the FS sends a 200 OK to this fine. At this > point the call is fine > > > > 30 minutes later they send a third re-invite because according > to them it is strictly for the purpose of ?keep alive? per RFC > 4028. This third re-invite has the exact same media IP and UDP > pot information as the second re-invite does. The problem is > FS does not respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the call to > be dropped as the other end does not recieve a response from > FS. > > > One more thing, we did not see the third re-invite in sofia > siptrace, but we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass > media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 17 08:13:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 10:13:41 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> Message-ID: <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> sip session timers is the standardized way to handle this. On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene wrote: > Are you doing proxy or bypass meda? > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Dec-09, at 12:53 AM, Juan Backson wrote: > > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls with > duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb > > On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: > >> Why not just set rtp-timeout-sec on the sofia profile and it'll do >> that for you. >> >> Unless something else is going on. >> >> /b >> >> On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: >> >> > Hi, >> > >> > I am having problem with around 1 % of the channels always get >> > zombilized. >> > >> > What I want to do is to have a background thread that regularly >> > check all the channels that have been in existance for like > 1 hr, >> > and then check to see if there is any RTP coming in and going out. >> > If there is no RTP, then I just hangup that channel. Does anyone >> > know if there is anyway to do that in a freeswitch module? Which >> > API can I use to accomplish this purpose? Alternatively, is there >> > anyway to configure freeswitch so that it will hangup the calls >> > where there is no media in and out for so many seconds? >> > >> > Thanks, >> > jb >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ddb3789f/attachment.html From stevesteffler at shaw.ca Thu Dec 17 08:13:15 2009 From: stevesteffler at shaw.ca (Steve Steffler) Date: Thu, 17 Dec 2009 09:13:15 -0700 Subject: [Freeswitch-users] mod_voicemail question In-Reply-To: <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> References: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> Message-ID: <1266CBCF-A246-4776-84CA-686CBCA93EA0@shaw.ca> Hello Micheal On Dec 15, 2009, at 12:09 PM, Michael Collins wrote: > Hi all, > > What is the difference between the mod_voicemail "vm_message_ext" parameter and the "file-extension" parameter? > > vm_message_ext is a channel variable: > http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext > > file-extension is a parameter of the voicemail module: > http://wiki.freeswitch.org/wiki/Mod_voicemail#file-extension > > The former sets for a specific user, the latter for mod_voicemail in general. Ahh, thanks for clearing this up for me! Now I understand the difference. > > I want all my voicemail in .WAV format except for a couple of extensions which need to be in MP3. > > I'm getting strange results playing with these settings, for example, after logging into the voicemail, it will say "You have 1 new message. First message at ", and then instead of the voicemail message there will be silence and a long pause. Then it will repeat the message count and loop this behavior. During the silence, I seem to be able to press keys to trigger voicemail events, like for example I am allowed to delete the message (although it isn't playing the message to me, and I am instead hearing silence). > > Any ideas? > > Is this perhaps a recording of silence, so that you might actually be listening to a message? > -MC Nope, turns out that according to the FS logs it was trying to play a {uuid}.WAV file that it was expecting to still be there, but which was deleted from a previous checking of voicemail. It's like the database of new messages was out of sync with the message sound files in the mailbox on the server. I deleted the 'ghost' voicemails from my mailbox and now things are back to normal. That could have been a result of my experimentations, I doubt it was a problem with FS. Thanks for your help and keep up the outstanding work! I love FreeSWITCH. :-) Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/50fd7f91/attachment.html From os at tenios.de Thu Dec 17 08:13:18 2009 From: os at tenios.de (=?iso-8859-1?Q?Oliver_Sch=F6nbeck?=) Date: Thu, 17 Dec 2009 17:13:18 +0100 Subject: [Freeswitch-users] Voicemail->Email Message-ID: <010101ca7f33$d9359eb0$8ba0dc10$@de> Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>&1 [[ -t 1 ]] && echo "Writing to logfile '$LOG'." exec > $LOG 2>&1 exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/971ab953/attachment.html From brian at freeswitch.org Thu Dec 17 08:17:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 10:17:15 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <010101ca7f33$d9359eb0$8ba0dc10$@de> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> Message-ID: <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: > Hello, > > we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. > > So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > typeset LOG="/tmp/${0##*/}.out" > mv $LOG ${LOG}.old >/dev/null 2>&1 > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > exec > $LOG 2>&1 > exim4 -t -v >> $LOG > > If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG > > Has anybody seen similar effects before? > > Any advice whats going wrong is heavily appreciated. > > Thanks > Oliver > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/556ebb4b/attachment-0001.html From djbinter at yahoo.com Thu Dec 17 08:19:32 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 08:19:32 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> Message-ID: <446963.62507.qm@web37507.mail.mud.yahoo.com> It only happened to the calls from this customer that keeps sending re-invite every 30 minutes, since their switch is expecting a reply back from those re-invite and FS did not respond back to those re-invite. Thank you.? ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:36:44 AM Subject: Re: [Freeswitch-users] SIP Re-invite if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. ?Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. ?Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: >? >Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >? >They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >? >30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/37dd7443/attachment.html From os at tenios.de Thu Dec 17 08:33:58 2009 From: os at tenios.de (=?iso-8859-1?Q?Oliver_Sch=F6nbeck?=) Date: Thu, 17 Dec 2009 17:33:58 +0100 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> Message-ID: <014201ca7f36$bbc371b0$334a5510$@de> Currently it is Version 1.0.trunk (15982) Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West Gesendet: Donnerstag, 17. Dezember 2009 17:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Voicemail->Email What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>&1 [[ -t 1 ]] && echo "Writing to logfile '$LOG'." exec > $LOG 2>&1 exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b54aad96/attachment.html From mayamatakeshi at gmail.com Thu Dec 17 08:50:48 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 18 Dec 2009 01:50:48 +0900 Subject: [Freeswitch-users] Small delay in registration validity In-Reply-To: <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> References: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> Message-ID: <15b9404e0912170850w348fb16at4dfafa7915207506@mail.gmail.com> On Fri, Dec 18, 2009 at 1:12 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The sql is sorted into transactions to boost performance so it waits for > either 500 statements to execute or 500ms to elapse to accumulate as many > sql stmts as possible into the transaction. > > set sql-in-transactions to false in your profile or make a patch to make > the 500ms configurable > Thanks. To change the param sql-in-transactions is enough for me (just during tests). I tested setting it to false and the behavior is as expected. I have updated the wiki: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#sql-in-transactions > On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote: > >> It seems to me, in previous revisions of FS, we could successfully call a >> registered user as soon as his terminal gets 200 OK for REGISTER. >> But after testing recent revisions, it seems we must wait a little (I wait >> 1 second) otherwise a call to bridge would end with this: >> >> 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create >> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> >> Similar thing is happening when the terminal unregisters: after >> unregistration an immediate call to bridge sofia/profile/user%domain will >> succeed. >> >> Has anything changed recently in the way registration works that could >> explain this? >> >> br, >> takeshi >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c289dc0c/attachment-0001.html From codecomplete at free.fr Thu Dec 17 08:56:30 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 08:56:30 -0800 (PST) Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? Message-ID: <26831043.post@talk.nabble.com> Hello I'm no wget expert, and figured I should ask here first: I'd like to download the whole wiki using wget for off-line reading. Using the following didn't work: wget -m -np http://wiki.freeswitch.org/wiki/Main_Page If I move the wiki/ directory to the root directory of my web server, and try to open http://localhost/wiki/Main_Page, FireFox tries to download the page with this dialog box: "You have chosen to open Main_Page which is a: application/octet-stream" I assume wget can do this, but I don't know enough. Has someone succeeded in downloading the whole wiki with wget and could give the right switches to use? Thank you. -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831043.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 17 09:00:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 11:00:52 -0600 Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? In-Reply-To: <26831043.post@talk.nabble.com> References: <26831043.post@talk.nabble.com> Message-ID: I would rather you not do that with wget you beat the hell out of the wiki resources... how often do you do this? I would try doing a printable version. /b On Dec 17, 2009, at 10:56 AM, Fred-145 wrote: > > Hello > > I'm no wget expert, and figured I should ask here first: I'd like to > download the whole wiki using wget for off-line reading. > > Using the following didn't work: > > wget -m -np http://wiki.freeswitch.org/wiki/Main_Page > > If I move the wiki/ directory to the root directory of my web server, and > try to open http://localhost/wiki/Main_Page, FireFox tries to download the > page with this dialog box: > > "You have chosen to open > Main_Page > which is a: application/octet-stream" > > I assume wget can do this, but I don't know enough. Has someone succeeded in > downloading the whole wiki with wget and could give the right switches to > use? > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3604e24b/attachment.html From stevendt at primrosebank.net Thu Dec 17 09:02:10 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 17 Dec 2009 17:02:10 -0000 Subject: [Freeswitch-users] Building on Windows Message-ID: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) 5. How do I specify which options (e.g., mod_flite, to be included iin the build. 6. How do I build the sounds etc. ? Regards Dave 5>os_unix.obj : warning LNK4221: no public symbols found; archive member will be inaccessible 1>filesys.c 6>..\..\pcre\pcre_ucp_searchfuncs.c(158) : warning C4018: '<' : signed/unsigned mismatch 6>..\..\pcre\pcre_ucp_searchfuncs.c(163) : warning C4018: '<=' : signed/unsigned mismatch 16>getenv.c 15>..\..\src\switch_console.c(553) : warning C4244: '=' : conversion from '__w64 int' to 'int', possible loss of data 15>..\..\src\switch_console.c(584) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data 15>switch_core_media_bug.c 15>..\..\src\switch_core_media_bug.c(178) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(221) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(222) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 18>nta_tag.c 21>c:\freeswitch src\freeswitch\libs\xmlrpc-c\src\xmlrpc_server_abyss.c(894) : warning C4047: 'initializing' : 'TOsSocket' differs in levels of indirection from 'void *' 31>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>mod_opal.cpp 86>c:\freeswitch src\freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error C1083: Cannot open include file: 'ptlib.h': No such file or directory -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/fcbacae3/attachment.html From codecomplete at free.fr Thu Dec 17 09:14:21 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 09:14:21 -0800 (PST) Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? In-Reply-To: References: <26831043.post@talk.nabble.com> Message-ID: <26831566.post@talk.nabble.com> I only tried once. Maybe someone used to wget could generate a PDF in case people need an offline copy? -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831566.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From andrew at hijacked.us Thu Dec 17 09:15:27 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 17 Dec 2009 12:15:27 -0500 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> Message-ID: <20091217171527.GA16380@hijacked.us> On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > Hi, > > I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. > Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. > What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. > > 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? > Tortoise SVN is fine and is probably the de-facto client for windows. > 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? > > 4. There was one fatal error in the build of mod_opal (missing file) > (Some examples of the warnings and the error are shown below :-) > Try with VS 2008 and see if they go away. > 5. How do I specify which options (e.g., mod_flite, to be included iin the build. > You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. > 6. How do I build the sounds etc. ? > The sounds are a subproject too IIRC. Andrew From vizentini at hotmail.com Thu Dec 17 09:18:55 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Thu, 17 Dec 2009 17:18:55 +0000 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl Message-ID: Hi,I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURLBut I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. And sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid Gateway! This is my configuration (overlook language details ) "
"+ ""+ ""+ ""+ ""+ " "+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ "
"+ ""; User id "test" is able to register and call other internal users In my sip_profiles/internal.xml I have: Can you help me with this issue? Thank youPaulo _________________________________________________________________ Keep your friends updated?even when you?re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b3e1f312/attachment-0001.html From djbinter at yahoo.com Thu Dec 17 09:35:27 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 09:35:27 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> Message-ID: <569367.65001.qm@web37503.mail.mud.yahoo.com> Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/d47e003d/attachment.html From mike at jerris.com Thu Dec 17 09:37:07 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 12:37:07 -0500 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: <7E9B67A4-A2C0-4429-B62C-6B89F2858444@jerris.com> I have not seen anyone mention it. Mike On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote: > I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... > > Thanks, __Yehavi: > > 2009/12/17 Michael Jerris > Its software, anything is possible with enough time and effort. > > Mike > > On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > > > After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). > > > > Any idea whether it is possible to program Freeswitch to support this draft? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3a68e92f/attachment.html From brian at freeswitch.org Thu Dec 17 09:44:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 11:44:15 -0600 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: References: Message-ID: <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> I'm going to guess you removed these lines from your profile: parse=true causes the profile to parse the domain looking for gateways and register them.. /b On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote: > Hi, > I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURL > But I am getting this warning: > 2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. > > And > > sofia status gateway MyGW > API CALL [sofia(status gateway MyGW)] output: > Invalid Gateway! > > > This is my configuration (overlook language details ) > > "
"+ > ""+ > ""+ > ""+ > ""+ > " "+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > > ""+ > ""+ > > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > "
"+ > ""; > > User id "test" is able to register and call other internal users > > In my sip_profiles/internal.xml I have: > > > > > > > > Can you help me with this issue? > > Thank you > Paulo > > > Keep your friends updated? even when you?re not signed in. _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/edfdb7a9/attachment-0001.html From djbinter at yahoo.com Thu Dec 17 09:47:07 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 09:47:07 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <569367.65001.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <569367.65001.qm@web37503.mail.mud.yahoo.com> Message-ID: <209520.17661.qm@web37504.mail.mud.yahoo.com> I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540 Thank you. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/e78cc4a1/attachment.html From dave at 3c.co.uk Thu Dec 17 09:50:22 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 10:50:22 -0700 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <556038.39248.qm@web37502.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <556038.39248.qm@web37502.mail.mud.yahoo.com> Message-ID: <1261072222.6396.76.camel@local.freepabx.com> Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave > The trace that I pasted on the pastebin was from our > analyzer,Tektronix spectra2 that was sitting between FS and customer. > I also had the FS sip trace on and compare with the trace from Spectra > when I found out about the 3rd re-invite was missing from FS. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 7:57:42 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > The question was: > > Are you doing the packet capture on the actual FS box using tshark or > tcpdump? > > > On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: > We have a customer that we are sending calls to off > the FS and here is the issue: > > > > Call is initially setup fine and they send a first > re-invite with media 0.0.0.0 to place the caller on > hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP > to cut through media and the FS sends a 200 OK to this > fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because > according to them it is strictly for the purpose of > ?keep alive? per RFC 4028. This third re-invite has > the exact same media IP and UDP pot information as the > second re-invite does. The problem is FS does not > respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a > response from FS. > > > One more thing, we did not see the third re-invite in > sofia siptrace, but we do see it in ethereal, which is > kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in > bypass media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn: +19193869900 +19193869900 > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yhding2003 at yahoo.ca Thu Dec 17 09:47:28 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Thu, 17 Dec 2009 09:47:28 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> Message-ID: <26831042.post@talk.nabble.com> Hi, If I configure data as following, why FS A "1001" call FS B "1003" failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26831042.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Thu Dec 17 10:08:37 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 17 Dec 2009 10:08:37 -0800 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: References: Message-ID: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> I found the issue with this. I did an svn checkout from the trunk, and then I did a local svn export to another local folder. For some reason, the svn export did not include the libs/openzap folder (which was not the case when I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap folder? Best Regards, Jerry -----Original Message----- From: Brian West [mailto:brian at freeswitch.org] Sent: Wednesday, December 16, 2009 2:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 Need siptrace with this type "sofia profile xxxx siptrace on" replace xxxx with your profile. /b On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote: > I upgraded to the latest 1.0.5pre9 and now if I try to call from an > internal phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" > reply. Below is the console loglevel 7 output. It says the > destination is out-of-order. I'm not sure what this means. Any help is appreciated. > > 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for > proxy > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy > [0] > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for > proxy > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy > [0] > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/5381 at 192.168.72.141:5060 > [e58e763f-7688-4600-aa70-481bbc359f58] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel > sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: > v=0 > o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4 > 192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0 > PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000/1 > a=ptime:20 > a=ptime:20 > > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> > CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> > CS_ROUTING > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING > 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing > Anonymous->93491028 in context default > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->unloop] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) > [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) > [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match > (PASS) [tod_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(open=true) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match > (FAIL) [holiday_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->Mediant1000] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) > [Mediant1000] > destination_number(93491028) =~ /^8(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->SangomaPRI] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) > [SangomaPRI] > destination_number(93491028) =~ /^9(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(effective_caller_id_number=425740${caller_id_number}) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE EXECUTE > sofia/internal/5381 at 192.168.72.141:5060 set(open=true) > 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] EXECUTE > sofia/internal/5381 at 192.168.72.141:5060 > set(effective_caller_id_number=4257405381) > 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET > [effective_caller_id_number]=[4257405381] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string > 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot > create outgoing channel of type [openzap] cause: > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 > Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. > Cause: DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup > sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [KILL] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to > INVITE > with: 502 > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to > sleep > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session > 1 > (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external > entities > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session > 1 > (sofia/internal/5381 at 192.168.72.141:5060) Ended > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close > Channel sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From mike at jerris.com Thu Dec 17 10:14:28 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 13:14:28 -0500 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <20091217171527.GA16380@hijacked.us> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> Message-ID: On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: >> Hi, >> >> I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? >> >> 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. >> Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. >> What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules stand > a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. >> >> 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? >> > Tortoise SVN is fine and is probably the de-facto client for windows. > make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] >> 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. >> >> 4. There was one fatal error in the build of mod_opal (missing file) >> (Some examples of the warnings and the error are shown below :-) >> > Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. >> 5. How do I specify which options (e.g., mod_flite, to be included iin the build. >> > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer >> 6. How do I build the sounds etc. ? >> > > The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/680983a0/attachment.html From brian at freeswitch.org Thu Dec 17 10:14:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 12:14:31 -0600 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> References: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> Message-ID: <8E309ACD-6284-4AE6-BF26-711E45125A5F@freeswitch.org> This would have nothing to do with receiving a 502 on sip. /b On Dec 17, 2009, at 12:08 PM, Jerry Richards wrote: > I found the issue with this. I did an svn checkout from the trunk, and then > I did a local svn export to another local folder. For some reason, the svn > export did not include the libs/openzap folder (which was not the case when > I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap > folder? > > Best Regards, > Jerry From djbinter at yahoo.com Thu Dec 17 10:31:56 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 10:31:56 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <1261072222.6396.76.camel@local.freepabx.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <556038.39248.qm@web37502.mail.mud.yahoo.com> <1261072222.6396.76.camel@local.freepabx.com> Message-ID: <926177.41302.qm@web37504.mail.mud.yahoo.com> Yes, I have a complete trace here: http://pastebin.freeswitch.org/11541 ________________________________ From: David Knell To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:50:22 AM Subject: Re: [Freeswitch-users] SIP Re-invite Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave > The trace that I pasted on the pastebin was from our > analyzer,Tektronix spectra2 that was sitting between FS and customer. > I also had the FS sip trace on and compare with the trace from Spectra > when I found out about the 3rd re-invite was missing from FS. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 7:57:42 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > The question was: > > Are you doing the packet capture on the actual FS box using tshark or > tcpdump? > > > On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: > We have a customer that we are sending calls to off > the FS and here is the issue: > > > > Call is initially setup fine and they send a first > re-invite with media 0.0.0.0 to place the caller on > hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP > to cut through media and the FS sends a 200 OK to this > fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because > according to them it is strictly for the purpose of > ?keep alive? per RFC 4028. This third re-invite has > the exact same media IP and UDP pot information as the > second re-invite does. The problem is FS does not > respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a > response from FS. > > > One more thing, we did not see the third re-invite in > sofia siptrace, but we do see it in ethereal, which is > kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in > bypass media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn: +19193869900 +19193869900 > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/07c5b91c/attachment-0001.html From yhding2003 at yahoo.ca Thu Dec 17 10:33:09 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Thu, 17 Dec 2009 10:33:09 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <26831042.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> Message-ID: <26832823.post@talk.nabble.com> param name="username" value="1101" param name="password" value="1234" param name="proxy" value="192.168.129.194:5060" param name="register" value="false" Hi, If I configure data as following, why FS A "1001" call FS B "1003" failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From djbinter at yahoo.com Thu Dec 17 10:53:30 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 10:53:30 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <569367.65001.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <569367.65001.qm@web37503.mail.mud.yahoo.com> Message-ID: <25799.44344.qm@web37505.mail.mud.yahoo.com> Please advise whether I should put a request in JIRA. http://pastebin.freeswitch.org/11541 Thank you. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/08f4f658/attachment.html From kristian.kielhofner at gmail.com Thu Dec 17 11:01:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 14:01:16 -0500 Subject: [Freeswitch-users] Handling REFER... Message-ID: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: ------------------------------------------------------------------------ REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: "NONAME" ;tag=BagvZeKSrj7yH From: ;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: Contact: Content-Length: 0 ------------------------------------------------------------------------ send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: ;tag=203332153_1430350929_10 To: "NONAME" ;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From juanbackson at gmail.com Thu Dec 17 11:05:14 2009 From: juanbackson at gmail.com (Juan Backson) Date: Fri, 18 Dec 2009 03:05:14 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> Message-ID: <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> Hi, I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true and minimum-session-expires=120. Is this the correct way of setting the sip session timers? thanks, jb On Fri, Dec 18, 2009 at 12:13 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > sip session timers is the standardized way to handle this. > > > > On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene wrote: > >> Are you doing proxy or bypass meda? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Dec-09, at 12:53 AM, Juan Backson wrote: >> >> Hi >> >> I have rtp-timeout-sec set to 300 s but I am still getting calls with >> duration of 1 day long. >> >> Is there any other ways to check for zombie channels? >> >> jb >> >> On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: >> >>> Why not just set rtp-timeout-sec on the sofia profile and it'll do >>> that for you. >>> >>> Unless something else is going on. >>> >>> /b >>> >>> On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: >>> >>> > Hi, >>> > >>> > I am having problem with around 1 % of the channels always get >>> > zombilized. >>> > >>> > What I want to do is to have a background thread that regularly >>> > check all the channels that have been in existance for like > 1 hr, >>> > and then check to see if there is any RTP coming in and going out. >>> > If there is no RTP, then I just hangup that channel. Does anyone >>> > know if there is anyway to do that in a freeswitch module? Which >>> > API can I use to accomplish this purpose? Alternatively, is there >>> > anyway to configure freeswitch so that it will hangup the calls >>> > where there is no media in and out for so many seconds? >>> > >>> > Thanks, >>> > jb >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> > users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/eefd9bd7/attachment-0001.html From brian at freeswitch.org Thu Dec 17 11:21:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 13:21:35 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> Message-ID: Please try on SVN trunk. I might toss a PRE10 sooner. /b On Dec 17, 2009, at 1:05 PM, Juan Backson wrote: > Hi, > > I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true and minimum-session-expires=120. > > Is this the correct way of setting the sip session timers? > > thanks, > jb From brian at proximosystems.com Thu Dec 17 11:29:14 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 14:29:14 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: <049601ca7f4f$37da5580$a78f0080$@com> Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7f7ba068/attachment.html From brian at freeswitch.org Thu Dec 17 11:33:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 13:33:42 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: If you're going to have that many listeners then it would be best to use something like shoutcast to broadcast the stream out to a local stream on various different boxes... then tie the callers into a stream... when they have questions uuid_transfer them into the conf.. then back to the stream when done. This would scale to very large numbers because you could split it out into 100's of boxes if needed. /b On Dec 17, 2009, at 1:29 PM, Brian wrote: > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I?m doing wrong, but I don?t see what it could be. > > Brian. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/c79a9440/attachment.html From yehavi.bourvine at gmail.com Thu Dec 17 11:36:40 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 21:36:40 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West > Also what device are you using? I haven't tested with many so far... > Polycom, Snom and a few others do TLS (see interop page on wiki) others do > it wrong. > > /b > > On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: > > You could try ssldump: > > http://www.rtfm.com/ssldump/ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/2ea8a523/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 17 11:42:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 13:42:03 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/4d726120/attachment.html From mike at jerris.com Thu Dec 17 11:43:10 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 14:43:10 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: We are always doing enhancements and yes there are some real scalability enhancements in trunk compared to 1.0.4, I am just not sure if they effect conference significantly or not. I would guess that trunk is actually more stable than 1.0.4 at the moment. Give it a try and find out. Mike On Dec 17, 2009, at 2:29 PM, Brian wrote: > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I?m doing wrong, but I don?t see what it could be. > > Brian. > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > I would be curious what the same tests produce with svn trunk of FreeSWITCH. > > Mike > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn?t scale nearly as well as I?d hoped (based on what I?ve read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I?ve set file logging to ?notice? level. My conference profile is configured to suppress several events, hoping that it would improve performance. > > Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: > > Scenario 1: > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/39907451/attachment-0001.html From steveu at coppice.org Thu Dec 17 11:50:08 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 18 Dec 2009 03:50:08 +0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: <4B2A8B70.20300@coppice.org> On 12/18/2009 03:29 AM, Brian wrote: > > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > Brian. > I don't think you have mentioned which codecs are involved. This can have a profound effect. Steve > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > Mike > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > From jeff at jefflenk.com Thu Dec 17 12:06:38 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 17 Dec 2009 12:06:38 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> Message-ID: <1261080398190-4183177.post@n2.nabble.com> The sounds projects (which do the downloads and extraction) are not present for 2005. Also alot of the newer modules dont have build support either. I would suggest you use VS2008 Express Michael Jerris wrote: > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > >> On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: >>> Hi, >>> >>> I'm probably going to regret this - I'm not sure that I'll be able to do >>> this without a lot of pain (nothing to do with FS - more my lack of >>> ability with Visual Studio), but......, I want to try building >>> FreeSwitch from source rather than using the pre-built binaries. I have >>> a couple of initial questions that, hopefully, someone can answer please >>> ? >>> >>> 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on >>> the horizon for me. >>> Having downloaded the SVN, I see there is a VS 2005 Solution, but it is >>> marked as "Unsupported", although the Wiki says that you only need >>> VC++2005. >>> What does "unsupported" mean in this context ? I guess that support for >>> VS2005 is being dropped, but is the VS2005 Solution still being >>> maintained, and if so, for how long? I'd hate to get into the build >>> thing and then find that I was stalled when VS2005 support was dropped >>> altogether ? >> >> Install VS 2008 if at all possible (express edition is free). 2005 >> support isn't maintained much if at all, so a lot of newer modules stand >> a good chance of not having support. > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > >>> >>> 2. The whole SVN thing is new to me but I've worked out that I need an >>> SVN Client on Windows to work with the source. Can anyone recommend the >>> best (free) SVN Client for Windows to use with FreeSwitch. I have >>> installed TortoiseSVN - a Windows Explorer Shell that looks pretty and >>> seemed to work on my first build but it's not command line based so some >>> of the tips given in the Wiki like "make current" and "make sounds" may >>> be more awkward to achieve. Is anyone else using Tortoise and/or can >>> give some tips on which SVN client to use ? >>> >> Tortoise SVN is fine and is probably the de-facto client for windows. >> > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] >>> 3. I built 15979 last night (with VS2005) and got some warnings, with >>> data type conversion - is this a known issue under Windows ? > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > >>> >>> 4. There was one fatal error in the build of mod_opal (missing file) >>> (Some examples of the warnings and the error are shown below :-) >>> >> Try with VS 2008 and see if they go away. > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > >>> 5. How do I specify which options (e.g., mod_flite, to be included iin >>> the build. >>> >> You can enable the different sub projects somehow in the UI, I always >> forget exactly how but just click around in VS and you'll find it. > > You can adjust this in the configuration managaer > >>> 6. How do I build the sounds etc. ? >>> >> >> The sounds are a subproject too IIRC. > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > Mike > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4183177.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Thu Dec 17 12:10:54 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 17 Dec 2009 12:10:54 -0800 (PST) Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> Message-ID: <1261080654294-4183200.post@n2.nabble.com> I run FreeSWITCH on a Windows Server 2008 R2 (x64) box with several analog lines and it works very well. mercutioviz wrote: > > And we shouldn't be using 1.0.4 anyway, should we? ;) > -MC > > > On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva > wrote: > >> I've been using FreeSWITCH on Windows lately and seems to work pretty >> well. >> Sangoma has been testing more and more lately the Windows drivers with >> FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 >> though, always using trunk, if you are going to be using PSTN lines (and >> therefore requiring openzap) I think it would be a good idea for you to >> use >> trunk and latest wanpipe drivers. >> >> -- >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >> 9T3 Canada >> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Windows-Stable-enough-for-production-use-tp4174199p4183200.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 17 12:14:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 14:14:58 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <4B2A8B70.20300@coppice.org> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <4B2A8B70.20300@coppice.org> Message-ID: Yes, while it is true that does make a profound difference but if he has many listeners and not very many talkers... just tapping into the conference and streaming that audio out would scale well. /b On Dec 17, 2009, at 1:50 PM, Steve Underwood wrote: > I don't think you have mentioned which codecs are involved. This can > have a profound effect. > > Steve From brian at proximosystems.com Thu Dec 17 12:32:23 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 15:32:23 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> Message-ID: <04b501ca7f58$0a188870$1e499950$@com> I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ae93dca3/attachment-0001.html From vizentini at hotmail.com Thu Dec 17 12:46:00 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Thu, 17 Dec 2009 20:46:00 +0000 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> References: , <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> Message-ID: Hi,FS was sending (while loading modules) such request: [purpose] => gateways But I was not aware of that...so that I am replying FS with my Gateways now... But now I am wondering...suppose I have 1000 domains and two different gateways per domain (2K Gateways) Should I reply FS request with such huge XML on startup? Thanks for your backings PauloFrom: brian at freeswitch.org Date: Thu, 17 Dec 2009 11:44:15 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl I'm going to guess you removed these lines from your profile: parse=true causes the profile to parse the domain looking for gateways and register them.. /b On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote:Hi,I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURLBut I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. And sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid Gateway! This is my configuration (overlook language details ) "
"+ ""+ ""+ ""+ ""+ " "+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ "
"+ ""; User id "test" is able to register and call other internal users In my sip_profiles/internal.xml I have: Can you help me with this issue? Thank youPaulo Keep your friends updated? even when you?re not signed in. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/40ac82d6/attachment.html From anthony.minessale at gmail.com Thu Dec 17 12:48:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 14:48:33 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04b501ca7f58$0a188870$1e499950$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> Message-ID: <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum to use > for this topic from now on. > > > > Thanks, > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 2:42 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > One man's stable release is another man's 6 month old release with hundreds > of known fixed bugs. > If one of the core developers tells you to try it, you may as well take the > time to try it now that you have opened a forum questioning the scalability. > > When you tested asterisk did you actually use 600 phones and verify that > each one can hear the audio perfectly and in time with what the speaker was > saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or follow any > of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have a > policy against entertaining load testing questions but if you like asterisk, > by all means, use it, and good luck to you if those numbers you are testing > at are what you plan to put in real production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3c69cc8a/attachment-0001.html From dave at 3c.co.uk Thu Dec 17 13:06:48 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 21:06:48 +0000 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04b501ca7f58$0a188870$1e499950$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> Message-ID: <1261084008.6396.89.camel@local.freepabx.com> Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Dec 17 13:07:25 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 15:07:25 -0600 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: References: , <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> Message-ID: <51C17D71-0193-40BF-9ED6-39A1BA58E5D6@freeswitch.org> In your case don't store them in the domain put them in the gateways tags on the profile directly. /b On Dec 17, 2009, at 2:46 PM, Paulo Vicentini wrote: > Hi, > FS was sending (while loading modules) such request: [purpose] => gateways > But I was not aware of that...so that I am replying FS with my Gateways now... > > But now I am wondering...suppose I have 1000 domains and two different gateways per domain (2K Gateways) > Should I reply FS request with such huge XML on startup? > > > Thanks for your backings > > Paulo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/46de59c3/attachment.html From Prometheus001 at gmx.net Thu Dec 17 13:20:15 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 17 Dec 2009 22:20:15 +0100 Subject: [Freeswitch-users] How to overcome 415 Unsupported Media Type Message-ID: <4B2AA08F.3000102@gmx.net> I try to attach Bravis video conference clients to Freeswitch. Those video conference clients are really working good (Multilingual clients for testing ca be downloaded here: http://www.bravis.eu/). Some big companies here in Germany use them in large installations. They are based on SIP, but do not use any publicly known codecs. Normally they are maintained and routed via our OpenSIPS server, but I would like to integrate them into our Freeswitch system. That way I do not have to manage 2 SIP servers for phone calls and video conferencing calls. However the SIP message does not provide Content-Type: application/sdp. Instead it provides Content-Type: application/BRAVIS. The clients register successfully but they do not invite. Freeswitch answers "SIP/2.0 415 Unsupported Media Type." I have added bypass_media=true into the dialplan and inbound-late-negotiation true in the internal profile but this didn't help. I think Freeswitch complains about the content-type. Is there any way how I may overcome this? Here is a sample Invite INVITE sip:835352 at sip5.mydomain.com SIP/2.0. From: myname ;tag=5c5c3ef6bbe9de119f1aa11f7ca41a5f. To: sip:835352 at sip5.mydomain.com. Via: SIP/2.0/UDP 217.xxx.xxx.xx6:5530;iid=9931;branch=z9hG4bKc4583ef6bbe9de119f1aa11f7ca41a5f;uas-addr=217.24.11.190;rport. CSeq: 4711 INVITE. Call-ID: 2-ee3d3ef6-bbe9-de11-9fa1-a11f7ca41a5f. Contact: "myname" . User-Agent: BRAVIS/1.5.20.27.4585 (Linux 2.6.31-16-generic; generic; Ubuntu 9.10; i686; de; 8). Max-Forwards: 70. Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS. Supported: 100rel. Content-Type: application/BRAVIS. Content-Length: 174. ACAABAAAFDAAABAAMEFBHCGLACAACAAACPKNBHGOAPLDABAAFAAADBABAAPPAAAAELAFAACAAAHDHCGGGMHIPPUPPPPPOPBEKHHHAPLDOPBEKHHHAPLDABAAAAAADCABAAADFBMDHOAEAAAAAAGIGPHDHEAAPPPPPPJFKGAPLHHNKF. Best regards Peter From ujjval at simplesignal.com Thu Dec 17 13:38:38 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 17 Dec 2009 13:38:38 -0800 Subject: [Freeswitch-users] Performance Tuning Message-ID: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following": Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one ....any references on Wiki? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/31ec13fc/attachment.html From brian at proximosystems.com Thu Dec 17 13:41:01 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 16:41:01 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> Message-ID: <04ca01ca7f61$a0560e30$e1022a90$@com> I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/5b0fa004/attachment-0001.html From vinuth.madinur at gmail.com Thu Dec 17 13:53:55 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Fri, 18 Dec 2009 03:23:55 +0530 Subject: [Freeswitch-users] Performance Tuning In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Message-ID: <910309030912171353v17372138idd3c8e1cb5b15ee1@mail.gmail.com> 1. http://wiki.freeswitch.org/wiki/Modules.conf.xml 2. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence 3. http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles Might not be entirely helpful, but basically you can use either the external or internal profiles and change the ports, etc., as required. 4. You can disable mod_cdr_csv and enable mod_xml_cdr based on #1. Thanks, Vinuth. On Fri, Dec 18, 2009 at 3:08 AM, Ujjval Karihaloo wrote: > Looking at Performance Tune my Freeswitch > > > > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > > > > > > > > Is refers to the following?: > > > > > > Turn off every module you don't need > > Turn presence off in the profiles > > libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles > > mod_cdr_csv is slower than mod_xml_cdr > > > > > > How do I change each one ?.any references on Wiki? > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/f710c9ed/attachment.html From brian at proximosystems.com Thu Dec 17 14:05:24 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 17:05:24 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261084008.6396.89.camel@local.freepabx.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <1261084008.6396.89.camel@local.freepabx.com> Message-ID: <04e901ca7f65$08edd830$1ac98890$@com> Hi Dave, That was one of the questions I had in my original post, was there an alternative way to implement a single speaker, many listener case? There was a suggestion proposed to use local streams instead of the conference. I'm not familiar with it, and I'm in the process of reading the wiki and source code to see what can be done with that. Thanks, Brian. -----Original Message----- From: David Knell [mailto:dave at 3c.co.uk] Sent: Thursday, December 17, 2009 4:07 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 17 14:06:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:06:04 -0600 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> Message-ID: <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> The calls inherit the context from the parent, I think there is a var you can set on the chan to pick what context to use in a transfer like transfer_context or something grep the code for it On Dec 17, 2009 1:07 PM, "Kristian Kielhofner" < kristian.kielhofner at gmail.com> wrote: Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: ------------------------------------------------------------------------ REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: "NONAME" >;tag=BagvZeKSrj7yH From: ;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: > Contact: Content-Length: 0 ------------------------------------------------------------------------ send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: ;tag=203332153_1430350929_10 To: "NONAME" >;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dc6db1a8/attachment.html From timuckun at gmail.com Thu Dec 17 14:13:33 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 18 Dec 2009 11:13:33 +1300 Subject: [Freeswitch-users] Performance Tuning In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Message-ID: <855e4dcf0912171413s36ed4453m1be00e454f89562a@mail.gmail.com> > > libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles If you only have one provider for your trunk is it possible to set up multiple profiles for enhanced performance? For example if I have multiple DDIs from the provider can I set up a different profile for each one? Or maybe based on some some sort of a pattern? From Prometheus001 at gmx.net Thu Dec 17 14:17:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 17 Dec 2009 23:17:00 +0100 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <010101ca7f33$d9359eb0$8ba0dc10$@de> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> Message-ID: <4B2AADDC.1030805@gmx.net> Hello Oliver, I have the same on Ubuntu wth newest trunk. Best regards Peter Oliver Sch?nbeck schrieb: > > Hello, > > > > we are running freeswitch 1.0.trunk and are currently trying to get > the mod_voicemail to send the received messages to the user by using > exim4 on a debian machine. > > > > So far we followed the instructions in the wiki article ( > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > > typeset LOG="/tmp/${0##*/}.out" > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > exec > $LOG 2>&1 > > exim4 -t -v >> $LOG > > > > If I run the script from the command line everything is working as > expected. If the script gets called by freeswitch I get the following > result in my logfile: > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > fault (core dumped) exim4 -t -v >> $LOG > > > > Has anybody seen similar effects before? > > > > Any advice whats going wrong is heavily appreciated. > > > > Thanks > > Oliver > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Dec 17 14:20:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 16:20:24 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: > I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. > > Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/47dfc698/attachment.html From anthony.minessale at gmail.com Thu Dec 17 14:46:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:46:03 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp On Thu, Dec 17, 2009 at 3:41 PM, Brian wrote: > I did a test with the trunk version for the one conference case, and it > is the same results as for 1.0.4. The audio failed at around 300 listeners. > Oddly though, it consumed less %CPU (240% instead of 300%), and yet the > audio still failed at the same number of listeners. > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 3:49 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > We didn't post it anywhere but we just get overwhelmed with them and many > of them are unfounded and take up a lot of time to track down. That does > not mean you have not found a real problem but the first step is trying > trunk. > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum to use > for this topic from now on. > > > > Thanks, > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 2:42 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > One man's stable release is another man's 6 month old release with hundreds > of known fixed bugs. > If one of the core developers tells you to try it, you may as well take the > time to try it now that you have opened a forum questioning the scalability. > > When you tested asterisk did you actually use 600 phones and verify that > each one can hear the audio perfectly and in time with what the speaker was > saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or follow any > of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have a > policy against entertaining load testing questions but if you like asterisk, > by all means, use it, and good luck to you if those numbers you are testing > at are what you plan to put in real production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/587668b6/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 17 14:47:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:47:01 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <4B2AADDC.1030805@gmx.net> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> <4B2AADDC.1030805@gmx.net> Message-ID: <191c3a030912171447o16c81a8bp34886d8ce4d923c5@mail.gmail.com> yah it's exim segfaulting because you have to configure it to emulate sendmail per the wiki page. On Thu, Dec 17, 2009 at 4:17 PM, Peter P GMX wrote: > Hello Oliver, > > I have the same on Ubuntu wth newest trunk. > > Best regards > Peter > > Oliver Sch?nbeck schrieb: > > > > Hello, > > > > > > > > we are running freeswitch 1.0.trunk and are currently trying to get > > the mod_voicemail to send the received messages to the user by using > > exim4 on a debian machine. > > > > > > > > So far we followed the instructions in the wiki article ( > > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > > > > > I added some lines to the bash script to enable some kind of logging: > > #! /bin/bash > > > > typeset LOG="/tmp/${0##*/}.out" > > > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > > > exec > $LOG 2>&1 > > > > exim4 -t -v >> $LOG > > > > > > > > If I run the script from the command line everything is working as > > expected. If the script gets called by freeswitch I get the following > > result in my logfile: > > > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > > fault (core dumped) exim4 -t -v >> $LOG > > > > > > > > Has anybody seen similar effects before? > > > > > > > > Any advice whats going wrong is heavily appreciated. > > > > > > > > Thanks > > > > Oliver > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/22c65448/attachment.html From kristian.kielhofner at gmail.com Thu Dec 17 15:59:45 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 18:59:45 -0500 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> Message-ID: <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> Thanks for the hint! force_transfer_context and force_transfer_dialplan. I've updated the wiki (I'll add an example once I test it). On Thu, Dec 17, 2009 at 5:06 PM, Anthony Minessale wrote: > The calls inherit the context from the parent, I think there is a var you > can set on the chan to pick what context to use in a transfer like > transfer_context or something grep the code for it > > On Dec 17, 2009 1:07 PM, "Kristian Kielhofner" > wrote: > > Hello everyone, > > I've got two profiles running: s2s and trunk. ?The context for s2s is > defined as s2s-in. ?The context for trunk is defined as trunk-in. > trunk is bound to 192.168.168.3. > > recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: > ? ------------------------------------------------------------------------ > ? REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 > ? To: "NONAME" ;tag=BagvZeKSrj7yH > ? From: > ;tag=203332153_1430350929_10 > ? Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac > ? CSeq: 2 REFER > ? Max-Forwards: 70 > ? Refer-To: > ? Contact: > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: > ? ------------------------------------------------------------------------ > ? SIP/2.0 202 Accepted > ? Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 > ? From: > ;tag=203332153_1430350929_10 > ? To: "NONAME" ;tag=BagvZeKSrj7yH > ? Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac > ? CSeq: 2 REFER > ? Contact: > ? User-Agent: FreeSWITCH > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > ? Supported: precondition, path, replaces > ? Allow-Events: talk, refer > ? Content-Length: 0 > > ?FS routed this to the s2s-in context, even though it was sent to the > trunk profile. ?Shouldn't it have ended up in trunk-in? ?For the time > being I wrote some crazy dialplan for s2s-in to transfer the call to > trunk-in but I'm wondering what could be going on here. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From frank at impactfax.com Thu Dec 17 16:01:50 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 17 Dec 2009 19:01:50 -0500 Subject: [Freeswitch-users] sip message logging and analysis Message-ID: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> I bit off topic but. Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/feb001e5/attachment.html From freeswitch at aastral.net Thu Dec 17 16:01:35 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 19:01:35 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> Message-ID: <4B2AC65F.5090806@aastral.net> Hey Brian, I've been doing some testing and I am unable to get auth-calls to work through a proxy the way I want them to, even with setting apply-proxy-acl to either the endpoint IP or the proxy IP. I have a multi-tenant system with multiple domains with multiple users in each domain. And I want to restrict a user to an arbitrary CIDR and challenge them for a password. The arbitrary CIDR will vary from UA to UA, and is specified in the directory via the auth-acl parameter. TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of the proxy. Thanks, Bill Brian West wrote: > it needs to be an ACL from acl.conf or a ip/cidr > > /b > > On Dec 17, 2009, at 5:41 AM, Bill W wrote: > >> Okay, I added: to my sofia >> profile and restarted sofia, and still no joy. >> >> I'm on FreeSWITCH Version 1.0.trunk (15764) >> I've got in >> the directory, but I'm still being rejected by the acl: >> >> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >> Rejected by user acl 190.218.103.12/32 >> >> Here's what I believe is the appropriate snippet of the debug output: >> http://pastebin.freeswitch.org/11531 >> >> Thoughts? >> Thanks, >> Bill > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Thu Dec 17 16:27:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 19:27:16 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact wrote: > I bit off topic but? > > > > Using FS to send calls sip to the LD carrier. > > > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call.? But this is? a > day later after the trouble has been reported. > > > > I am looking for guidance on how to log all sip message traffic and then be > able to easily retrieve to find a call and look at what sip messages really > were being based and by whom.? Maybe store them in a database or some other > file that might be opened by an analysis tool. > > > > Any suggestions on how to log this information and then what tool to use for > later analysis? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Thu Dec 17 16:27:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Dec 2009 16:27:58 -0800 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> Message-ID: <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> On Thu, Dec 17, 2009 at 3:59 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Thanks for the hint! > > force_transfer_context and force_transfer_dialplan. > > I've updated the wiki (I'll add an example once I test it). > > I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dba64367/attachment-0001.html From msc at freeswitch.org Thu Dec 17 16:33:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Dec 2009 16:33:11 -0800 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <87f2f3b90912171633v4ae885afv873cba3aa9cde23d@mail.gmail.com> On Thu, Dec 17, 2009 at 4:01 PM, Frank @ Impact wrote: > I bit off topic but? > > > > Using FS to send calls sip to the LD carrier. > > > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call. But this is aday later after the trouble has been reported. > > > > I am looking for guidance on how to log all sip message traffic and then be > able to easily retrieve to find a call and look at what sip messages really > were being based and by whom. Maybe store them in a database or some > other file that might be opened by an analysis tool. > > > > Any suggestions on how to log this information and then what tool to use > for later analysis? > > > Jason Garland's ClueCon2009 videos about tcpdump and wireshark cover the thought of doing a rotating log file so that it captures a bunch of stuff but doesn't go over X number of megabytes... I don't recall exactly where in his videos that part appears, but here are the links to those vids. Hope it helps! -MC Look at this video first: http://www.viddler.com/explore/cluecon/videos/33/ Then check this one if you need more info: http://www.viddler.com/explore/cluecon/videos/8/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dc372834/attachment.html From chris at fowler.cc Thu Dec 17 16:33:34 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 17 Dec 2009 19:33:34 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> I'm using VQManager (there is a 30 day trial) and it's useful for seeing who does what / when per call; it's very easy to install... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank @ Impact Sent: Thursday, December 17, 2009 4:02 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] sip message logging and analysis I bit off topic but... Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/82660632/attachment.html From brian at freeswitch.org Thu Dec 17 16:54:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 18:54:04 -0600 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> Message-ID: <99531772-685B-4CB0-AA48-E2F023A35366@freeswitch.org> Also when can we expect little KK's running around? :P Congrats on the marriage!!!! /b On Dec 17, 2009, at 6:27 PM, Michael Collins wrote: > I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. > > -MC From brian at freeswitch.org Thu Dec 17 16:54:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 18:54:44 -0600 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> Message-ID: <3CA5BAB4-A966-41F9-BAB6-4C4ED91CBA03@freeswitch.org> So is wireshark UI and its free! :P /b On Dec 17, 2009, at 6:33 PM, Chris Fowler wrote: > I?m using VQManager (there is a 30 day trial) and it?s useful for seeing who does what / when per call; it?s very easy to install? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/64fe187d/attachment.html From david.villasmil.work at gmail.com Thu Dec 17 16:54:49 2009 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 18 Dec 2009 01:54:49 +0100 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> Message-ID: <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by "," have fun! David El 18/12/2009, a las 01:27, Kristian Kielhofner escribi?: > Frank, > > Probably the cleanest (albeit non-FreeSWITCH) way to implement this > would be to use OpenSIPS/SER/etc between you and the carrier with the > siptrace module. > > But that's probably more work than you want. There's always tcpdump > with a decent filter (udp port 5060 and host x.x.x.x) and then > something like http://www.badpenguin.co.uk/files/pcap-util2 > > Both will allow you to search for BYEs and who is sending them. > > Also keep in mind that they (or you) may just be dropping the RTP > without ever sending a BYE. Setting the various RTP timeouts in > FreeSWITCH can help with that. You can then look for logs/events (are > there any for RTP timeout?) to see who's dropping RTP. > > On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact > wrote: >> I bit off topic but? >> >> >> >> Using FS to send calls sip to the LD carrier. >> >> >> >> Some calls have problems where they drop the call or audio drops or >> whatever. >> >> The carrier?s first response is that we dropped the call. But thi >> s is a >> day later after the trouble has been reported. >> >> >> >> I am looking for guidance on how to log all sip message traffic and >> then be >> able to easily retrieve to find a call and look at what sip >> messages really >> were being based and by whom. Maybe store them in a database or >> some other >> file that might be opened by an analysis tool. >> >> >> >> Any suggestions on how to log this information and then what tool >> to use for >> later analysis? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Thu Dec 17 17:21:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 18 Dec 2009 09:21:25 +0800 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> Message-ID: <23f91030912171721o6d4b9917ree41ec3092fb8a96@mail.gmail.com> I'm using contrib/seven/sip/sip2db.rb 2009/12/18 David Villasmil : > i agree with christian, though i would use tshark. you can actually > get the fields you want (method and callid) and store them in a dB. > then you need to match them with a query. it is simple but Lots of work. > > look into -e and -E of tshark separate the fields by "," > > have fun! > > David > > El 18/12/2009, a las 01:27, Kristian Kielhofner ?> escribi?: > >> Frank, >> >> ?Probably the cleanest (albeit non-FreeSWITCH) way to implement this >> would be to use OpenSIPS/SER/etc between you and the carrier with the >> siptrace module. >> >> ?But that's probably more work than you want. ?There's always tcpdump >> with a decent filter (udp port 5060 and host x.x.x.x) and then >> something like http://www.badpenguin.co.uk/files/pcap-util2 >> >> ?Both will allow you to search for BYEs and who is sending them. >> >> ?Also keep in mind that they (or you) may just be dropping the RTP >> without ever sending a BYE. ?Setting the various RTP timeouts in >> FreeSWITCH can help with that. ?You can then look for logs/events (are >> there any for RTP timeout?) to see who's dropping RTP. >> >> On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact >> wrote: >>> I bit off topic but? >>> >>> >>> >>> Using FS to send calls sip to the LD carrier. >>> >>> >>> >>> Some calls have problems where they drop the call or audio drops or >>> whatever. >>> >>> The carrier?s first response is that we dropped the call. ?But thi >>> s is ?a >>> day later after the trouble has been reported. >>> >>> >>> >>> I am looking for guidance on how to log all sip message traffic and >>> then be >>> able to easily retrieve to find a call and look at what sip >>> messages really >>> were being based and by whom. ?Maybe store them in a database or >>> some other >>> file that might be opened by an analysis tool. >>> >>> >>> >>> Any suggestions on how to log this information and then what tool >>> to use for >>> later analysis? >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Thu Dec 17 17:36:50 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 18 Dec 2009 09:36:50 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <26832823.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> <26832823.post@talk.nabble.com> Message-ID: <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> I couldn't guess what you want, pastbin your full config and logs and give more detail of your story perhaps someone can help you. 2009/12/18 yvonne ding : > > param name="username" value="1101" > param name="password" value="1234" > param name="proxy" value="192.168.129.194:5060" > param name="register" value="false" > > > Hi, > > If I configure data as following, why FS A "1001" call FS B "1003" failed ? > Thank you! > > FS A: 192.168.129.168, DN=1001 > FS B: 192.168.129.194, DN=1003 > > In FS A add /conf/sip_proifles/external/gwfsa.xml > ? > ? ? > > > > > ? ? > > > 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have > 1101 number > > > > > > Dan Le wrote: >> >> If you want FS server A to be able to call FS server B, you can set up a >> user account in server B's FS directory configs, and then just treat >> server >> B as a normal gateway by adding a gateway definition in server A. That >> will >> allow you to route calls to server B from A; to do the reverse, just >> mirror >> the configs the other direction. >> >> On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: >> >>> >>> I like to connect two freeswitch, call each other, communicate and vice >>> versa. >>> Can you give me an example for that? >>> >>> Thanks >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Dec 17 18:43:28 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 21:43:28 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AC65F.5090806@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> Message-ID: <4B2AEC50.3030305@metik.com> This may be difficult considering that ACL needs to consider the original src IP/URI. To do that it, freeswitch would need to do so using a header that retains that information (i.e. From, Via, Contact, etc.). Which I do not believe is currently possible using auth-acl or apply-proxy-acl. However, you should be able to emulate the behavior using mod_xml_curl (and validating against appropriate variables available when using it to authenticate the request). see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization -metik Bill W wrote: > Hey Brian, > > > I've been doing some testing and I am unable to get auth-calls to work > through a proxy the way I want them to, even with setting > apply-proxy-acl to either the endpoint IP or the proxy IP. > > I have a multi-tenant system with multiple domains with multiple users > in each domain. And I want to restrict a user to an arbitrary CIDR and > challenge them for a password. The arbitrary CIDR will vary from UA to > UA, and is specified in the directory via the auth-acl parameter. > > TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of > the proxy. > > > Thanks, > Bill > > Brian West wrote: > >> it needs to be an ACL from acl.conf or a ip/cidr >> >> /b >> >> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >> >> >>> Okay, I added: to my sofia >>> profile and restarted sofia, and still no joy. >>> >>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>> I've got in >>> the directory, but I'm still being rejected by the acl: >>> >>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>> Rejected by user acl 190.218.103.12/32 >>> >>> Here's what I believe is the appropriate snippet of the debug output: >>> http://pastebin.freeswitch.org/11531 >>> >>> Thoughts? >>> Thanks, >>> Bill >>> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Thu Dec 17 18:56:26 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 18:56:26 -0800 (PST) Subject: [Freeswitch-users] Creating Default Accounts on Directory Message-ID: <26838457.post@talk.nabble.com> Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at freeswitch.org Thu Dec 17 19:08:03 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 18 Dec 2009 01:08:03 -0200 Subject: [Freeswitch-users] Creating Default Accounts on Directory In-Reply-To: <26838457.post@talk.nabble.com> References: <26838457.post@talk.nabble.com> Message-ID: Please check your dialplan to match the new extension. You are looking for dialplan/default.xml extension Local_Extension. Check the cond destination_number, it should give you a good hint. Regards, JM On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz wrote: > > Hi Sir, > > I want to create a new xml file on the default directory of freeswitch > where 1000.xml is located, sample i created 9387821.xml and copy the > contents of the 1000.xml. > > The problem is when I used the account 9387821.xml and call 1000.xml it > doesn't work the message in freeswitch it always CS_DESTROY... Please help > me this with issue thanks... > > Edmar > > > -- > View this message in context: > http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/722a566d/attachment.html From freeswitch-users-list at metik.com Thu Dec 17 19:42:14 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 22:42:14 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <4B2AFA16.8020200@metik.com> Some providers do retain call data for diagnostic purposes and to to aid in troubleshooting. Why not politely ask them if they could provide you with a sip trace themselves or forward along the evidence that supported their conclusion. They should be willing to help you solve a problem that may potentially be of benefit to their other customers that report similar issues. Otherwise, as others suggest, you could simply capture the signaling and media traffic from the FS box itself using "tcpdump" (e.g. tcpdump -i eth0 -s 0 -w debug.pcap host 127.0.0.1 ) or ngrep (-d eth0 -W byline -O /tmp/debug.pcap host 127.0.0.1) and analyze the resulting file in Wirehark (Statistics->Voip Calls or Telephony->Voip Calls in the current version). If your provider is using a session border controller or does not have a distributed architecture, then you can replace 127.0.0.1 with the appropriate address. If not, then simply don't use the host filter at all (it will result in a larger capture file). I would just keep in mind that if an upstream device (NAT router, firewall, etc.) is wreaking havoc with session refreshes by dropping re-INVITEs or UPDATEs (associated with session refreshing), you may not see them because of your vantage point. The reason I typically recommend using the "-i" (tcpdump) and "-d" (ngrep) switch is to avoid linux 'cooked' captures (more of a personal preference since I occasionally do have to convert or merge captures). If you only have SSH access to your FS box, you may want to use tcpdump or ngrep along with "screen". "tshark" (tty/cli vesion of Wireshark) and "sipgrep" are also extremely useful. The later requires ngrep and a couple perl modules but I believe it is included with FS in the contrib or scripts directory--I forget which). -metik Frank @ Impact wrote: > > I bit off topic but? > > Using FS to send calls sip to the LD carrier. > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call. But this is > a day later after the trouble has been reported. > > I am looking for guidance on how to log all sip message traffic and > then be able to easily retrieve to find a call and look at what sip > messages really were being based and by whom. Maybe store them in a > database or some other file that might be opened by an analysis tool. > > Any suggestions on how to log this information and then what tool to > use for later analysis? > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From darklion11 at yahoo.com Thu Dec 17 19:53:23 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 19:53:23 -0800 (PST) Subject: [Freeswitch-users] Creating Default Accounts on Directory In-Reply-To: References: <26838457.post@talk.nabble.com> Message-ID: <26838750.post@talk.nabble.com> Hi Sir, Not working i set this to to call 8000001.xml up to 8000009.xml on the dialplan/default.xml same thing... Thanks, Edmar Jo?o Mesquita-4 wrote: > > Please check your dialplan to match the new extension. > > You are looking for dialplan/default.xml extension Local_Extension. Check > the cond destination_number, it should give you a good hint. > > Regards, > > JM > > On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz wrote: > >> >> Hi Sir, >> >> I want to create a new xml file on the default directory of >> freeswitch >> where 1000.xml is located, sample i created 9387821.xml and copy the >> contents of the 1000.xml. >> >> The problem is when I used the account 9387821.xml and call 1000.xml >> it >> doesn't work the message in freeswitch it always CS_DESTROY... Please >> help >> me this with issue thanks... >> >> Edmar >> >> >> -- >> View this message in context: >> http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838750.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 17 20:02:48 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 23:02:48 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AEC50.3030305@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> Message-ID: <4B2AFEE8.5020002@aastral.net> Hey Metik, Thanks for the reply, and the pointers for doing it with xml_curl. I'll guess have to do that in the short term, but in my opinion, having auth-acl be able to work through a proxy is very important as it is a vital part of a comprehensive security feature set. And it would be much simpler to implement from an end-user perspective than the alternative of doing it in xml_curl. As a matter of fact, I'm considering offering a bounty for that feature. What is the going rate for that kind of thing? Is anyone out there interested in coding this feature? Or chipping in for the bounty? Thanks, Bill Metik wrote: > This may be difficult considering that ACL needs to consider the > original src IP/URI. To do that it, freeswitch would need to do so > using a header that retains that information (i.e. From, Via, Contact, > etc.). Which I do not believe is currently possible using auth-acl or > apply-proxy-acl. > > However, you should be able to emulate the behavior using mod_xml_curl > (and validating against appropriate variables available when using it to > authenticate the request). > > see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization > > -metik > > > Bill W wrote: >> Hey Brian, >> >> >> I've been doing some testing and I am unable to get auth-calls to work >> through a proxy the way I want them to, even with setting >> apply-proxy-acl to either the endpoint IP or the proxy IP. >> >> I have a multi-tenant system with multiple domains with multiple users >> in each domain. And I want to restrict a user to an arbitrary CIDR and >> challenge them for a password. The arbitrary CIDR will vary from UA to >> UA, and is specified in the directory via the auth-acl parameter. >> >> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >> the proxy. >> >> >> Thanks, >> Bill >> >> Brian West wrote: >> >>> it needs to be an ACL from acl.conf or a ip/cidr >>> >>> /b >>> >>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>> >>> >>>> Okay, I added: to my sofia >>>> profile and restarted sofia, and still no joy. >>>> >>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>> I've got in >>>> the directory, but I'm still being rejected by the acl: >>>> >>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>> Rejected by user acl 190.218.103.12/32 >>>> >>>> Here's what I believe is the appropriate snippet of the debug output: >>>> http://pastebin.freeswitch.org/11531 >>>> >>>> Thoughts? >>>> Thanks, >>>> Bill >>>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users-list at metik.com Thu Dec 17 20:21:22 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 23:21:22 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AFEE8.5020002@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> Message-ID: <4B2B0342.3000201@metik.com> Why not simply implement this feature in the PROXY itself? FS has a pretty comprehensive security feature set for endpoints that directly register with it. Don't get me wrong, I do agree this is useful especially if you are going to be using your proxies to load balance across multiple FS boxes to create an ad-hoc cluster. I actually have session border controllers that have this feature and use it quite often. -metik Bill W wrote: > Hey Metik, > > Thanks for the reply, and the pointers for doing it with xml_curl. > > I'll guess have to do that in the short term, but in my opinion, having > auth-acl be able to work through a proxy is very important as it is a > vital part of a comprehensive security feature set. And it would be > much simpler to implement from an end-user perspective than the > alternative of doing it in xml_curl. > > As a matter of fact, I'm considering offering a bounty for that feature. > What is the going rate for that kind of thing? > > Is anyone out there interested in coding this feature? Or chipping in > for the bounty? > > > Thanks, > Bill > > > Metik wrote: > >> This may be difficult considering that ACL needs to consider the >> original src IP/URI. To do that it, freeswitch would need to do so >> using a header that retains that information (i.e. From, Via, Contact, >> etc.). Which I do not believe is currently possible using auth-acl or >> apply-proxy-acl. >> >> However, you should be able to emulate the behavior using mod_xml_curl >> (and validating against appropriate variables available when using it to >> authenticate the request). >> >> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >> >> -metik >> >> >> Bill W wrote: >> >>> Hey Brian, >>> >>> >>> I've been doing some testing and I am unable to get auth-calls to work >>> through a proxy the way I want them to, even with setting >>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>> >>> I have a multi-tenant system with multiple domains with multiple users >>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>> challenge them for a password. The arbitrary CIDR will vary from UA to >>> UA, and is specified in the directory via the auth-acl parameter. >>> >>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>> the proxy. >>> >>> >>> Thanks, >>> Bill >>> >>> Brian West wrote: >>> >>> >>>> it needs to be an ACL from acl.conf or a ip/cidr >>>> >>>> /b >>>> >>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>> >>>> >>>> >>>>> Okay, I added: to my sofia >>>>> profile and restarted sofia, and still no joy. >>>>> >>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>> I've got in >>>>> the directory, but I'm still being rejected by the acl: >>>>> >>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>> Rejected by user acl 190.218.103.12/32 >>>>> >>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>> http://pastebin.freeswitch.org/11531 >>>>> >>>>> Thoughts? >>>>> Thanks, >>>>> Bill >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mrene_lists at avgs.ca Thu Dec 17 22:08:16 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 18 Dec 2009 01:08:16 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AFEE8.5020002@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> Message-ID: <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> From looking at sofia.c, if the ip address of the caller is in apply- proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, and use that one for authentication. Is that what you did in your previous tests? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Dec-09, at 11:02 PM, Bill W wrote: > Hey Metik, > > Thanks for the reply, and the pointers for doing it with xml_curl. > > I'll guess have to do that in the short term, but in my opinion, > having > auth-acl be able to work through a proxy is very important as it is a > vital part of a comprehensive security feature set. And it would be > much simpler to implement from an end-user perspective than the > alternative of doing it in xml_curl. > > As a matter of fact, I'm considering offering a bounty for that > feature. > What is the going rate for that kind of thing? > > Is anyone out there interested in coding this feature? Or chipping in > for the bounty? > > > Thanks, > Bill > > > Metik wrote: >> This may be difficult considering that ACL needs to consider the >> original src IP/URI. To do that it, freeswitch would need to do so >> using a header that retains that information (i.e. From, Via, >> Contact, >> etc.). Which I do not believe is currently possible using auth-acl or >> apply-proxy-acl. >> >> However, you should be able to emulate the behavior using >> mod_xml_curl >> (and validating against appropriate variables available when using >> it to >> authenticate the request). >> >> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >> >> -metik >> >> >> Bill W wrote: >>> Hey Brian, >>> >>> >>> I've been doing some testing and I am unable to get auth-calls to >>> work >>> through a proxy the way I want them to, even with setting >>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>> >>> I have a multi-tenant system with multiple domains with multiple >>> users >>> in each domain. And I want to restrict a user to an arbitrary >>> CIDR and >>> challenge them for a password. The arbitrary CIDR will vary from >>> UA to >>> UA, and is specified in the directory via the auth-acl parameter. >>> >>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>> not of >>> the proxy. >>> >>> >>> Thanks, >>> Bill >>> >>> Brian West wrote: >>> >>>> it needs to be an ACL from acl.conf or a ip/cidr >>>> >>>> /b >>>> >>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>> >>>> >>>>> Okay, I added: to >>>>> my sofia >>>>> profile and restarted sofia, and still no joy. >>>>> >>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>> I've got >>>> param> in >>>>> the directory, but I'm still being rejected by the acl: >>>>> >>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>> 64.135.119.105 >>>>> Rejected by user acl 190.218.103.12/32 >>>>> >>>>> Here's what I believe is the appropriate snippet of the debug >>>>> output: >>>>> http://pastebin.freeswitch.org/11531 >>>>> >>>>> Thoughts? >>>>> Thanks, >>>>> Bill >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Thu Dec 17 22:30:32 2009 From: freeswitch at aastral.net (Bill W) Date: Fri, 18 Dec 2009 01:30:32 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2B0342.3000201@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> Message-ID: <4B2B2188.2060803@aastral.net> Hey Metik, That's exactly what I'm trying to do... load balance across multiple FS boxes, and have any machine in the cluster be able to reach a device behind a NAT firewall. Hence the need for the proxy. Also, I'm trying to keep the proxy relatively "dumb" and put all the logic in the FS boxes. True I could do the auth on the proxies as well, but then I'm setting up another authentication scheme in addition to what is on the FS boxes, and then integrating the databases so everything is consistent. I also have hosts that talk to the FS boxes directly, rather than through the proxy. So I can't get rid of auth_acl on FS either, even if I do implement it on the proxies. So my setup becomes much more complex and potentially brittle. And all we're really talking about for FreeSWITCH, conceptually speaking, is populating a variable with a different IP. We could even make it configurable, as to which IP is to be used for the auth-acl. What are you using for SBCs? (if you are allowed to divulge that) I'm currently using OpenSIPS for my proxy. Thanks, Bill Metik wrote: > Why not simply implement this feature in the PROXY itself? > > FS has a pretty comprehensive security feature set for endpoints that > directly register with it. > > Don't get me wrong, I do agree this is useful especially if you are > going to be using your proxies to load balance across multiple FS boxes > to create an ad-hoc cluster. I actually have session border controllers > that have this feature and use it quite often. > > -metik > > Bill W wrote: >> Hey Metik, >> >> Thanks for the reply, and the pointers for doing it with xml_curl. >> >> I'll guess have to do that in the short term, but in my opinion, having >> auth-acl be able to work through a proxy is very important as it is a >> vital part of a comprehensive security feature set. And it would be >> much simpler to implement from an end-user perspective than the >> alternative of doing it in xml_curl. >> >> As a matter of fact, I'm considering offering a bounty for that feature. >> What is the going rate for that kind of thing? >> >> Is anyone out there interested in coding this feature? Or chipping in >> for the bounty? >> >> >> Thanks, >> Bill >> >> >> Metik wrote: >> >>> This may be difficult considering that ACL needs to consider the >>> original src IP/URI. To do that it, freeswitch would need to do so >>> using a header that retains that information (i.e. From, Via, Contact, >>> etc.). Which I do not believe is currently possible using auth-acl or >>> apply-proxy-acl. >>> >>> However, you should be able to emulate the behavior using mod_xml_curl >>> (and validating against appropriate variables available when using it to >>> authenticate the request). >>> >>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>> >>> -metik >>> >>> >>> Bill W wrote: >>> >>>> Hey Brian, >>>> >>>> >>>> I've been doing some testing and I am unable to get auth-calls to work >>>> through a proxy the way I want them to, even with setting >>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>> >>>> I have a multi-tenant system with multiple domains with multiple users >>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>> UA, and is specified in the directory via the auth-acl parameter. >>>> >>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>> the proxy. >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> Brian West wrote: >>>> >>>> >>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>> >>>>> /b >>>>> >>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>> >>>>> >>>>> >>>>>> Okay, I added: to my sofia >>>>>> profile and restarted sofia, and still no joy. >>>>>> >>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>> I've got in >>>>>> the directory, but I'm still being rejected by the acl: >>>>>> >>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>> Rejected by user acl 190.218.103.12/32 >>>>>> >>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>> http://pastebin.freeswitch.org/11531 >>>>>> >>>>>> Thoughts? >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Thu Dec 17 23:19:26 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 23:19:26 -0800 (PST) Subject: [Freeswitch-users] Destination Formats Expression Message-ID: <26840010.post@talk.nabble.com> Hi Everyone, Is there a link or tutorial for the expressions format. Example: 10 - default number [01[ - second number that start only on 0 or 1; [0-9] - 0 to 9 can be use Is there any? Thanks, Edmar -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840010.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Thu Dec 17 23:34:11 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 18 Dec 2009 18:34:11 +1100 Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <26840010.post@talk.nabble.com> References: <26840010.post@talk.nabble.com> Message-ID: <20091218073411.GA32288@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there a link or tutorial for the expressions format. Anything that describes Perl regular expressions should help, and for reference, see the pcre(3) manual page, and use the pcretest program to experiment. From lei.tlfly at gmail.com Thu Dec 17 23:42:46 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 18 Dec 2009 15:42:46 +0800 Subject: [Freeswitch-users] how does FS failover or load balance outbound calls between tow proxy Message-ID: <50c41b4e0912172342y67c4c8b2h1a99e41407b7eaaf@mail.gmail.com> Hi All I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load balance and failover to FS cluster by OpenSips, It works well. The problem is, the outbound calls from FS must also route throw then OpenSIPS servers. So, does FS servers can loadbalance the outbound calls between the two OpenSIPS servers and failover if one of the Opensips server is down? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/443ea268/attachment.html From msc at freeswitch.org Thu Dec 17 23:51:26 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 17 Dec 2009 23:51:26 -0800 Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <20091218073411.GA32288@jdc.jasonjgw.net> References: <26840010.post@talk.nabble.com> <20091218073411.GA32288@jdc.jasonjgw.net> Message-ID: <840F0EA0-227F-4CCD-BCB8-2F4945205A70@freeswitch.org> On Dec 17, 2009, at 11:34 PM, Jason White wrote: > Edmar Cruz wrote: >> >> Is there a link or tutorial for the expressions format. > > Anything that describes Perl regular expressions should help, and for > reference, see the pcre(3) manual page, and use the pcretest program > to > experiment. > http://wiki.freeswitch.org/wiki/Regular_Expression -MC From darklion11 at yahoo.com Thu Dec 17 23:58:17 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 23:58:17 -0800 (PST) Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <20091218073411.GA32288@jdc.jasonjgw.net> References: <26840010.post@talk.nabble.com> <20091218073411.GA32288@jdc.jasonjgw.net> Message-ID: <26840254.post@talk.nabble.com> Thanks that will be a great help Jason White-14 wrote: > > Edmar Cruz wrote: >> >> Is there a link or tutorial for the expressions format. > > Anything that describes Perl regular expressions should help, and for > reference, see the pcre(3) manual page, and use the pcretest program to > experiment. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840254.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From djbinter at yahoo.com Fri Dec 18 00:10:29 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 00:10:29 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> Message-ID: <922386.16417.qm@web37502.mail.mud.yahoo.com> Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/7ac70921/attachment-0001.html From devel at thom.fr.eu.org Fri Dec 18 01:51:08 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Fri, 18 Dec 2009 10:51:08 +0100 Subject: [Freeswitch-users] Voicemail->Email Message-ID: I get the same result with sendmail. This used to work in 1.0.3 , and after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is still there. Fran?ois On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Sch?nbeck wrote: Currently it is Version 1.0.trunk (15982) VON: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] IM AUFTRAG VON Brian West GESENDET: Donnerstag, 17. Dezember 2009 17:17 AN: freeswitch-users at lists.freeswitch.org BETREFF: Re: [Freeswitch-users] Voicemail->Email What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>"> [[ -t 1 ]] &">exec > $LOG 2>">exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [1] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [2] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [3] Links: ------ [1] mailto:FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/3292d179/attachment.html From Prometheus001 at gmx.net Fri Dec 18 02:00:55 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 18 Dec 2009 11:00:55 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: <4B28D6FD.6010702@gmx.net> References: <4B28D6FD.6010702@gmx.net> Message-ID: <4B2B52D7.9030505@gmx.net> Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: > Hello, > > we have the following scenario: > A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For > the called FS user, call forwarding has been enabled to another PSTN > extension (B) . > Result: The calling party does not hear any ringing tone. Here an > Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact > the same Patton Gateway): > > PSTN(A)====INVITE===>FS > PSTN(A)<===TRYING===>FS > FS===INVITE==>PSTN(B) > FS<==TRYING===PSTN(B) > FS<==RINGING==PSTN(B) > PSTN(A)<==PROGRESS===FS > FS<===OK======PSTN(B) > FS====ACK====>PSTN(B) > PSTN(A)<===OK========FS > PSTN(A)====ACK======>FS > > I would expect that FS answers RINGING back to PSTN(A). Instead it only > answers SESSION PROGRESS. > When PSTN(B) answers, they can hear each other, but there was no ringing > tone to PSTN(A) before. > > Are there any hints to overcome this, besides playing early media to > PSTN(A)? > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From viper at fx-services.com Fri Dec 18 02:21:12 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 18 Dec 2009 11:21:12 +0100 Subject: [Freeswitch-users] LUA and return variables Message-ID: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> Hi guys (and girls)! I'm working on a little bit of ENUM trickery and I tried doing some (illegal) nested conditions. :-) What I want to do is to first check enum with the ENUM application, then depending on the answer do some stuff. Say that the domain part of the ENUM answer is robin.nl, then I want to do action X instead of just briding the enum answer directly as I see in most examples. But I remembered that it wasn't allowed to do nested conditions. So what I did was stacked conditions. After that I read the dialplan wiki pages again and figured that my regexp never matches because variables I "set" during some phase of the extension I can't use in the same "go" as another condition. So, now my plan is to use LUA to do the regexp. I'll pass the enum answer to a lua script which will split the answer in a user and domain part and return those two to the main app. Then based on those two vars I'll do routing or other actions (like, prefix and then route). Is this how I'm supposed to do it? I can't find many examples on manipulating ENUM answers, other than bridging them directly. I can't change the way I do stuff to ENUM answers, because in most cases I'll just route them out the standard way. Anyone with experience on fiddling with ENUM answers? -- Robin Vleij From bcxml at hotmail.com Fri Dec 18 03:16:12 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 18 Dec 2009 03:16:12 -0800 (PST) Subject: [Freeswitch-users] Ringing after call has been rejected Message-ID: <26842055.post@talk.nabble.com> I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Fri Dec 18 05:04:51 2009 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Dec 2009 13:04:51 +0000 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> Message-ID: Brian, You haven't said what codecs are being used yet. Are the listeners using a different codec to the speaker? If so, you're potentially doing transcoding on every single channel, which would make CPU usage skyrocket. -Steve 2009/12/17 Anthony Minessale : > What exactly is your test process? > > you should try increasing the interval in the conference profile to a bigger > time slice maybe 30 40 or 60ms > you could also increase the ptime to match as well. > > > like brian said you could use mod_shout to broadcast the single speaker to > icecast and let people listen with itunes/winamp > > > On Thu, Dec 17, 2009 at 3:41 PM, Brian wrote: >> >> I did a test with the trunk version for the one conference case, and it is >> the same results as for 1.0.4. The audio failed at around 300 listeners. >> Oddly though, it consumed less %CPU (240% instead of 300%), and yet the >> audio still failed at the same number of listeners. >> >> >> >> Brian. >> >> >> >> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] >> Sent: Thursday, December 17, 2009 3:49 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> We didn't post it anywhere but we just get overwhelmed with them and many >> of them are unfounded and take up a lot of time to track down.? That does >> not mean you have not found a real problem but the first step is trying >> trunk. >> >> >> On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: >> >> I didn?t realize there was a policy about load testing questions. What >> forum should I have used for this? >> >> >> >> I didn?t get the chance to test on FS trunk yet, but when I do I will >> provide you with the feedback when I do. Just let me know what forum to use >> for this topic from now on. >> >> >> >> Thanks, >> >> >> >> Brian. >> >> >> >> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] >> Sent: Thursday, December 17, 2009 2:42 PM >> >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> One man's stable release is another man's 6 month old release with >> hundreds of known fixed bugs. >> If one of the core developers tells you to try it, you may as well take >> the time to try it now that you have opened a forum questioning the >> scalability. >> >> When you tested asterisk did you actually use 600 phones and verify that >> each one can hear the audio perfectly and in time with what the speaker was >> saying?? Did you try same on FS? >> >> Did you optimize your dialplan on FS to deal with a load test or follow >> any of the recommended performance tuning page. >> >> All of the answers to these questions are really moot because we have a >> policy against entertaining load testing questions but if you like asterisk, >> by all means, use it, and good luck to you if those numbers you are testing >> at are what you plan to put in real production......... >> >> On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: >> >> Hi Mike, >> >> >> >> I didn?t get around to testing on the FreeSWITCH trunk yet. Are there >> substantial fixes to mod_conference in the FreeSWITCH trunk that might >> increase capacity for my scenario of one speaker and many listeners? If I >> want to put this into a production environment, I would need a stable >> version, which as far as I know is the 1.0.4 version. >> >> >> >> However, I did test on Asterisk 1.4 using app_conference, and doing the >> same scenario was able to get 1 speaker and 600 listeners on a single >> conference with no audio issues. The CPU at that point was just over 300%, >> same as where the single conference scenario failed on FreeSWITCH with 300 >> listeners. ?I was able to push it to over 700 listeners before I reached >> 400% CPU usage (I guess maxing out my quad-core processors), and asterisk >> finally crashed. But up until that point, there were no audio problems. >> >> >> >> I?ve read a lot about how FreeSWITCH is supposed to be more scalable than >> Asterisk, but unless there is something wrong with my FreeSWITCH setup, >> Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH >> capacity in this case. Again, maybe there is something on the FreeSWITCH >> side that I?m doing wrong, but I don?t see what it could be. >> >> >> >> Brian. >> >> >> >> >> >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Thursday, December 17, 2009 10:18 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> I would be curious what the same tests produce with svn trunk of >> FreeSWITCH. >> >> >> >> Mike >> >> >> >> On Dec 16, 2009, at 4:49 PM, Brian wrote: >> >> >> >> Hi, >> >> >> >> I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to >> see if it will scale better that other solutions. My scenario is to have one >> speaker, and many listeners (mute). Since I have only one speaker, I was >> expecting this to scale well because there is no audio mixing required, just >> send each frame of the single speaker to each listener. Unfortunately, my >> testing was disappointing, and it didn?t scale nearly as well as I?d hoped >> (based on what I?ve read on how FreeSWITCH is supposed to be generally very >> scalable). >> >> >> >> Here?s my server setup is this: >> >> >> >> FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of >> RAM. I?ve set file logging to ?notice? level. My conference profile is >> configured to suppress several events, hoping that it would improve >> performance. >> >> >> >> Here are a few scenarios I tested, and roughly where I reached the point >> of audio failure on the conferences: >> >> >> >> Scenario 1: >> >> 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) >> >> >> >> Scenario 2: >> >> 4 conferences, 1 speaker per conference, audio failed approx 110 listeners >> per conference (so just over 400 total channels on the system). >> >> >> >> Scenario 3: >> >> 16 conferences, 1 speaker per conference, audio failed at 32 listeners per >> conference (so just over 500 total channels on the system). >> >> >> >> >> >> Looking at the output from ?top?, it seems that in all 3 scenarios, the >> audio quality failed when the % CPU for the FreeSWITCH process exceeded >> 300%. >> >> >> >> I was hoping maybe someone else might have done similar testing, or maybe >> has suggestions on how to improve the performance. Or perhaps an alternate >> solution to the one speaker, many listener case? >> >> >> >> Thanks, >> >> >> >> Brian. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From keith at laaks.com Fri Dec 18 05:13:48 2009 From: keith at laaks.com (Keith Laaks) Date: Fri, 18 Dec 2009 15:13:48 +0200 (SAST) Subject: [Freeswitch-users] mod_xml_ldap compile issue. Message-ID: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> Hi, I am having an issue getting mod_xml_ldap to compile properly.... making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 ------------------------------------------------------------------------ I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? Best Regards Keith From brian at proximosystems.com Fri Dec 18 06:08:31 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 09:08:31 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> Message-ID: <007101ca7feb$9462de70$bd289b50$@com> I've got FS running on a 64 bit OS, and here is more info on the test procedure. I've got one server (primary) that hosts the speaker call (this is meant to be a primary conference with a few speakers, but my test simplifies this to just one speaker). I've got a second server (secondary) that hosts the conference that all the listeners go into, and I have two other servers that I use automate the listener calls. The goal is to have several secondary servers to scale the listener side of things, but for this initial test I've only got one secondary server. The primary server dials into the secondary conference server so that the listeners can hear the speaker conference on the primary server. The automated listener servers start dialing into the listener conference at a combined rate of 5 calls per second (i.e. 2.5 calls per second each). The play an audio loop that represents noise on their end, which since they are listeners, should be ignored anyway. As I ramp up the automated listener calls, I manually call into the conference from either my SIP phone, or from a land line using a DID that I have directed to the conference. All calls are using SIP with uLaw 8000hz codec. Also, I've set up the profile for the listener conference to disable many of the events: I do have caller controls for the listener, since in my production I will need to generate and handle events for listener DTMF. To compare FreeSWITCH vs Asterisk, I just swap out the secondary conference server and everything else stays the same. Brian. From: Brian West [mailto:brian at freeswitch.org] Sent: Thursday, December 17, 2009 5:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/8845d8d1/attachment.html From brian at freeswitch.org Fri Dec 18 06:43:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Dec 2009 08:43:27 -0600 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: <4B2B52D7.9030505@gmx.net> References: <4B28D6FD.6010702@gmx.net> <4B2B52D7.9030505@gmx.net> Message-ID: That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. /b On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: > Should I open a JIRA for this? > > Best regards > Peter From yhding2003 at yahoo.ca Fri Dec 18 06:54:32 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Fri, 18 Dec 2009 06:54:32 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> <26832823.post@talk.nabble.com> <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> Message-ID: <26844589.post@talk.nabble.com> Hi, As far as I know, there are two ways to connect two freeswitch, by using ACL or using authentication. Also from this email history discussion, another solution is to create user in FS B directory,then treat server B as normal gateway by adding gateway definiton in FS A. So my question is how to connect FS A and FS B through ACL or through the way this email described. The information I pasted is about the last way. FS A: 192.168.129.168, caller id= 1001 FS B: 192.168.129.194, callee id= 1003, create 1101 for gateway configure In FS A add /conf/sip_proifles/external/gwfsa.xml param name="username" value="1101" param name="password" value="1234" param name="proxy" value="192.168.129.194:5060" param name="register" value="false" note: I delete < and /> for param cause it can't be displayed in this email. Both FS A and FS B are default configuration except creating id=1101 on FS B side. I'm confused if I connect two freeswitch by using ACLs, How do I confiugre data in both side ? Your kind help is highly appreciated. Seven Du wrote: > > I couldn't guess what you want, pastbin your full config and logs and > give more detail of your story perhaps someone can help you. > > 2009/12/18 yvonne ding : >> >> param name="username" value="1101" >> param name="password" value="1234" >> param name="proxy" value="192.168.129.194:5060" >> param name="register" value="false" >> >> >> Hi, >> >> If I configure data as following, why FS A "1001" call FS B "1003" failed >> ? >> Thank you! >> >> FS A: 192.168.129.168, DN=1001 >> FS B: 192.168.129.194, DN=1003 >> >> In FS A add /conf/sip_proifles/external/gwfsa.xml >> ? >> ? ? >> >> >> >> >> ? ? >> >> >> 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't >> have >> 1101 number >> >> >> >> >> >> Dan Le wrote: >>> >>> If you want FS server A to be able to call FS server B, you can set up a >>> user account in server B's FS directory configs, and then just treat >>> server >>> B as a normal gateway by adding a gateway definition in server A. That >>> will >>> allow you to route calls to server B from A; to do the reverse, just >>> mirror >>> the configs the other direction. >>> >>> On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz >>> wrote: >>> >>>> >>>> I like to connect two freeswitch, call each other, communicate and vice >>>> versa. >>>> Can you give me an example for that? >>>> >>>> Thanks >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> View this message in context: >> http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26844589.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jbr at consiglia.dk Fri Dec 18 07:23:27 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 18 Dec 2009 16:23:27 +0100 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: I have found some ways to get presence, or rather BLF functions to work on Snom telephones in a distributed network with several FSs. I'll post a solution on the wiki when I have tested it further. Anyhow, I'm using the mod_event_multicast module with the following configuration: With this setting on all FSs, the registration table is also automatically updated thus listing all sets registered across all FSs. In the table sip_registrations (under the database for the profile used), the field status has the value: "Registered" if the UA is registered on another FS and the value "Registered(UDP)" if the UA is registered on the same FS. The field server_host, however, is the ip-address of "local" FS. Now comes the question: is there any way to let the field server_host show the server address of the server actually registered to? Or any other way using the existing modules to get the information about which FS the UAs are registered to? The information is going to be used for the routing decisions between networked FSs. /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/fbe8a447/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Fri Dec 18 07:29:35 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Fri, 18 Dec 2009 16:29:35 +0100 Subject: [Freeswitch-users] mod_xml_ldap compile issue. In-Reply-To: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> References: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> Message-ID: <4B2B9FDF.9060600@puzzled.xs4all.nl> On 12/18/2009 02:13 PM, Keith Laaks wrote: > Hi, > > I am having an issue getting mod_xml_ldap to compile properly.... > > > making all mod_xml_cdr > > making all mod_xml_ldap > Creating mod_xml_ldap.la... > /usr/bin/ld: > /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): > relocation R_X86_64_32S against `.rodata' can not be used when making a > shared object; recompile with -fPIC > /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: > could not read symbols: Bad value > collect2: ld returned 1 exit status > cat: .libs/mod_xml_ldap.log: No such file or directory > make[5]: *** [mod_xml_ldap.la] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_xml_ldap-all] Error 1 > make[2]: *** [all-recursive] Error 1 > ------------------------------------------------------------------------ > I notice the openldap library has been bumped up to .19 - not sure if that > may have anything to do with it. > > At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. > > mod_ldap compiles OK, but mod_xml_ldap fails as per the above. > > What am I doing working here ? I had the same issue and MikeJ (one of the core developers) looked at it. Conclusion was that it is an openldap issue and iirc the solution is to libtoolize libraries/liblutil/Makefile.in so that when running configure a Makefile with proper compiler flags is generated in libraries/liblutil/ Patches welcome :) Regards, Patrick From mike at jerris.com Fri Dec 18 07:29:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Dec 2009 10:29:13 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <007101ca7feb$9462de70$bd289b50$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> <007101ca7feb$9462de70$bd289b50$@com> Message-ID: What is your dialplan on the secondary box? On Dec 18, 2009, at 9:08 AM, Brian wrote: > I?ve got FS running on a 64 bit OS, and here is more info on the tes > t procedure. > > > > I?ve got one server (primary) that hosts the speaker call (this is m > eant to be a primary conference with a few speakers, but my test sim > plifies this to just one speaker). I?ve got a second server (seconda > ry) that hosts the conference that all the listeners go into, and I > have two other servers that I use automate the listener calls. The g > oal is to have several secondary servers to scale the listener side > of things, but for this initial test I?ve only got one secondary ser > ver. > > > > The primary server dials into the secondary conference server so > that the listeners can hear the speaker conference on the primary > server. > > > > The automated listener servers start dialing into the listener > conference at a combined rate of 5 calls per second (i.e. 2.5 calls > per second each). The play an audio loop that represents noise on > their end, which since they are listeners, should be ignored anyway. > > > > As I ramp up the automated listener calls, I manually call into the > conference from either my SIP phone, or from a land line using a DID > that I have directed to the conference. > > > > All calls are using SIP with uLaw 8000hz codec. Also, I?ve set up th > e profile for the listener conference to disable many of the events: > > > > > > > > > > > > > > > > > > > > I do have caller controls for the listener, since in my production I > will need to generate and handle events for listener DTMF. > > > > To compare FreeSWITCH vs Asterisk, I just swap out the secondary > conference server and everything else stays the same. > > > > Brian. > > > > From: Brian West [mailto:brian at freeswitch.org] > Sent: Thursday, December 17, 2009 5:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > What exactly are you doing I know it goes better than that.. are you > using 64bit? > > > > / b > > > > On Dec 17, 2009, at 3:41 PM, Brian wrote: > > > > > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c45449a5/attachment.html From freeswitch at aastral.net Fri Dec 18 07:53:11 2009 From: freeswitch at aastral.net (Bill W) Date: Fri, 18 Dec 2009 10:53:11 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> Message-ID: <4B2BA567.6010202@aastral.net> Hello Mathieu, I assumed that apply-proxy-acl was a modifier of auth-calls, so in my quick tests I just hard-coded the UA IP in the profile. And I get: 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.97.83/32 Where 64.135.119.105 is the IP of my proxy. And actually this is a REGISTER, not an INVITE. I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the register packet. I will be incommunicado for the rest of today, but when I get back online, I'll see if I can get my proxy to add the X-AUTH-IP to the REGISTER packet and see if that makes a difference. Thanks for your help! Bill Mathieu Rene wrote: > From looking at sofia.c, if the ip address of the caller is in apply- > proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, > and use that one for authentication. > Is that what you did in your previous tests? > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Dec-09, at 11:02 PM, Bill W wrote: > >> Hey Metik, >> >> Thanks for the reply, and the pointers for doing it with xml_curl. >> >> I'll guess have to do that in the short term, but in my opinion, >> having >> auth-acl be able to work through a proxy is very important as it is a >> vital part of a comprehensive security feature set. And it would be >> much simpler to implement from an end-user perspective than the >> alternative of doing it in xml_curl. >> >> As a matter of fact, I'm considering offering a bounty for that >> feature. >> What is the going rate for that kind of thing? >> >> Is anyone out there interested in coding this feature? Or chipping in >> for the bounty? >> >> >> Thanks, >> Bill >> >> >> Metik wrote: >>> This may be difficult considering that ACL needs to consider the >>> original src IP/URI. To do that it, freeswitch would need to do so >>> using a header that retains that information (i.e. From, Via, >>> Contact, >>> etc.). Which I do not believe is currently possible using auth-acl or >>> apply-proxy-acl. >>> >>> However, you should be able to emulate the behavior using >>> mod_xml_curl >>> (and validating against appropriate variables available when using >>> it to >>> authenticate the request). >>> >>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>> >>> -metik >>> >>> >>> Bill W wrote: >>>> Hey Brian, >>>> >>>> >>>> I've been doing some testing and I am unable to get auth-calls to >>>> work >>>> through a proxy the way I want them to, even with setting >>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>> >>>> I have a multi-tenant system with multiple domains with multiple >>>> users >>>> in each domain. And I want to restrict a user to an arbitrary >>>> CIDR and >>>> challenge them for a password. The arbitrary CIDR will vary from >>>> UA to >>>> UA, and is specified in the directory via the auth-acl parameter. >>>> >>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>>> not of >>>> the proxy. >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> Brian West wrote: >>>> >>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>> >>>>> /b >>>>> >>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>> >>>>> >>>>>> Okay, I added: to >>>>>> my sofia >>>>>> profile and restarted sofia, and still no joy. >>>>>> >>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>> I've got >>>>> param> in >>>>>> the directory, but I'm still being rejected by the acl: >>>>>> >>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>>> 64.135.119.105 >>>>>> Rejected by user acl 190.218.103.12/32 >>>>>> >>>>>> Here's what I believe is the appropriate snippet of the debug >>>>>> output: >>>>>> http://pastebin.freeswitch.org/11531 >>>>>> >>>>>> Thoughts? >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Fri Dec 18 07:55:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 18 Dec 2009 10:55:52 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2BA567.6010202@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> <4B2BA567.6010202@aastral.net> Message-ID: <11616732-C3B1-4B03-9368-2C777C402F1F@avgs.ca> You need to add that header manually in your OpenSIPS config, FreeSWITCH wont look in record-route/via to try to guess it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 18-Dec-09, at 10:53 AM, Bill W wrote: > Hello Mathieu, > > I assumed that apply-proxy-acl was a modifier of auth-calls, so in my > quick tests I just hard-coded the UA IP in the profile. > > > > > And I get: > 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP > 64.135.119.105 > Rejected by user acl 190.218.97.83/32 > > Where 64.135.119.105 is the IP of my proxy. And actually this is a > REGISTER, not an INVITE. > > I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the > register > packet. > > I will be incommunicado for the rest of today, but when I get back > online, I'll see if I can get my proxy to add the X-AUTH-IP to the > REGISTER packet and see if that makes a difference. > > > Thanks for your help! > Bill > > > Mathieu Rene wrote: >> From looking at sofia.c, if the ip address of the caller is in apply- >> proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, >> and use that one for authentication. >> Is that what you did in your previous tests? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Dec-09, at 11:02 PM, Bill W wrote: >> >>> Hey Metik, >>> >>> Thanks for the reply, and the pointers for doing it with xml_curl. >>> >>> I'll guess have to do that in the short term, but in my opinion, >>> having >>> auth-acl be able to work through a proxy is very important as it >>> is a >>> vital part of a comprehensive security feature set. And it would be >>> much simpler to implement from an end-user perspective than the >>> alternative of doing it in xml_curl. >>> >>> As a matter of fact, I'm considering offering a bounty for that >>> feature. >>> What is the going rate for that kind of thing? >>> >>> Is anyone out there interested in coding this feature? Or chipping >>> in >>> for the bounty? >>> >>> >>> Thanks, >>> Bill >>> >>> >>> Metik wrote: >>>> This may be difficult considering that ACL needs to consider the >>>> original src IP/URI. To do that it, freeswitch would need to do so >>>> using a header that retains that information (i.e. From, Via, >>>> Contact, >>>> etc.). Which I do not believe is currently possible using auth- >>>> acl or >>>> apply-proxy-acl. >>>> >>>> However, you should be able to emulate the behavior using >>>> mod_xml_curl >>>> (and validating against appropriate variables available when using >>>> it to >>>> authenticate the request). >>>> >>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>> >>>> -metik >>>> >>>> >>>> Bill W wrote: >>>>> Hey Brian, >>>>> >>>>> >>>>> I've been doing some testing and I am unable to get auth-calls to >>>>> work >>>>> through a proxy the way I want them to, even with setting >>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>> >>>>> I have a multi-tenant system with multiple domains with multiple >>>>> users >>>>> in each domain. And I want to restrict a user to an arbitrary >>>>> CIDR and >>>>> challenge them for a password. The arbitrary CIDR will vary from >>>>> UA to >>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>> >>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>>>> not of >>>>> the proxy. >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> Brian West wrote: >>>>> >>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>> >>>>>> /b >>>>>> >>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>> >>>>>> >>>>>>> Okay, I added: to >>>>>>> my sofia >>>>>>> profile and restarted sofia, and still no joy. >>>>>>> >>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>> I've got >>>>>> param> in >>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>> >>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>>>> 64.135.119.105 >>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>> >>>>>>> Here's what I believe is the appropriate snippet of the debug >>>>>>> output: >>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>> >>>>>>> Thoughts? >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Dec 18 07:56:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 09:56:02 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: References: Message-ID: <191c3a030912180756i613e67d4ha5fef2142a70f82d@mail.gmail.com> oh really, sendmail segfaults? if another application is crashing you need to figure that out, whatever used to work doesnt now so you need to figure out what it was and let us know. On Fri, Dec 18, 2009 at 3:51 AM, Fran?ois Legal wrote: > I get the same result with sendmail. This used to work in 1.0.3 , and after > upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is > still there. > > > > Fran?ois > > > > On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Sch?nbeck wrote: > > Currently it is Version 1.0.trunk (15982) > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Brian West > *Gesendet:* Donnerstag, 17. Dezember 2009 17:17 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] Voicemail->Email > > > > What SVN rev. exactly? > > > > /b > > > > On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: > > > > Hello, > > > > we are running freeswitch 1.0.trunk and are currently trying to get the > mod_voicemail to send the received messages to the user by using exim4 on a > debian machine. > > > > So far we followed the instructions in the wiki article ( > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > > typeset LOG="/tmp/${0##*/}.out" > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > exec > $LOG 2>&1 > > exim4 -t -v >> $LOG > > > > If I run the script from the command line everything is working as > expected. If the script gets called by freeswitch I get the following result > in my logfile: > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > fault (core dumped) exim4 -t -v >> $LOG > > > > Has anybody seen similar effects before? > > > > Any advice whats going wrong is heavily appreciated. > > > > Thanks > > Oliver > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/93835bd7/attachment.html From fdelawarde at wirelessmundi.com Fri Dec 18 08:12:08 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 18 Dec 2009 17:12:08 +0100 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <1261152728.11815.57.camel@luna.tc.commsmundi.com> Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) Fran?ois. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 3:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > We didn't post it anywhere but we just get overwhelmed with them and > many of them are unfounded and take up a lot of time to track down. > That does not mean you have not found a real problem but the first > step is trying trunk. > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Dec 18 08:34:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 10:34:21 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261152728.11815.57.camel@luna.tc.commsmundi.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> Message-ID: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:888 at conference.freeswitch.orgThis is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > > substantial fixes to mod_conference in the FreeSWITCH trunk that might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > > than Asterisk, but unless there is something wrong with my FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other solutions. My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the single > > speaker to each listener. Unfortunately, my testing was disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > > of RAM. I?ve set file logging to ?notice? level. My conference profile > > is configured to suppress several events, hoping that it would improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/8ba43e55/attachment-0001.html From msc at freeswitch.org Fri Dec 18 08:54:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 08:54:23 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting Shortly! Message-ID: <87f2f3b90912180854h59b651c2t289f5a42ebec3973@mail.gmail.com> Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll get into the agenda. Talk to you all soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/df3e27df/attachment.html From anthony.minessale at gmail.com Fri Dec 18 09:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 11:21:19 -0600 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <26842055.post@talk.nabble.com> References: <26842055.post@talk.nabble.com> Message-ID: <191c3a030912180921t4f188291w5f36ff3eb647f924@mail.gmail.com> not answering it would be the best way. if you want to generate fake congestion you can use tone_stream:// or gentones On Fri, Dec 18, 2009 at 5:16 AM, bcxml wrote: > > I have an incomming call being answered by FreeSwitch and passed to IVR > application which rejects the call. > > The call is never answered by FreeSwitch, but instead of hearing a busy > signal, the caller hears ringing. > > Can anyone advise how I can get the user to hear a busy signal after call > rejection instead of ringing. > > Here is the debug trace > > http://pastebin.freeswitch.org/11558 > > Thanks > > > Brian > > -- > View this message in context: > http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/60f3ddf7/attachment.html From srinivas.ksvreddy at gmail.com Thu Dec 17 23:41:08 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Fri, 18 Dec 2009 13:11:08 +0530 Subject: [Freeswitch-users] Fwd: incoming call In-Reply-To: References: Message-ID: Hi, i have up the freeswitch with domain(eg sipserver.domain.com) name instead of local ip, two clints are regitered with freeswitch using domain name(eg sipserver.domain.com), one client is making a call to other one, other clint receiving a invite request like this 173927 3120.658532 10.91.154.108 10.91.154.80 SIP/SDP Request: INVITE sip:1010 at 10.91.154.80:5061, with session description, but usually it should come with INVITE sip:1010 at sipserver.domain.com? what is changes i need to do for this? any idea? Regards-- Srinivasula Reddy K -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/ca16ea09/attachment.html From mike at jerris.com Fri Dec 18 09:37:16 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Dec 2009 12:37:16 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <922386.16417.qm@web37502.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> Message-ID: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: > Mike, > > My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > Thank you, > Dorn B. > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 8:03:46 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > are you doing this trace from the freeswitch box itself? > > Mike > > On Dec 17, 2009, at 10:48 AM, DJB wrote: > >> Anthony, >> >> I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >> Please advise if you need further info. >> >> Thank you. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/3a50f5f6/attachment.html From fdelawarde at wirelessmundi.com Fri Dec 18 09:41:44 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 18 Dec 2009 18:41:44 +0100 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Message-ID: <1261158104.11815.92.camel@luna.tc.commsmundi.com> It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (in the more tested sense) than openzap+mod_openzap. I just use both projects separately or together depending on what's needed! I'm no banker nor do I understand the code, but many thanks for all those unpaid contributions providing an excellent alternative for free telephony. Your names really deserve being engraved in google's cache for eternity. :-) But still, I would like to see those numbers... Fran?ois. On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to sponsor > the effort to add features to the conference module, but of course, I > do not hold my breath, instead I continue to improve it for free when > I find time. This is one of many reasons I do not enjoy performance > discussions unless I am talking to an engineer who understands the > code or a banker ready to pay for improvements. That is not my way of > saying pay me or forget it as you can clearly see the conference > module has made it to where it is today with no financial support at > all. Just the efforts of myself and several brave volunteers over the > years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > wrote: > Hearing that Asterisk (1.4) scales 2x like FS is not common, > sounds like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk > version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see > who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference > case, and > > it is the same results as for 1.0.4. The audio failed at > around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead > of 300%), > > and yet the audio still failed at the same number of > listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with > them and > > many of them are unfounded and take up a lot of time to > track down. > > That does not mean you have not found a real problem but the > first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > > wrote: > > > > I didn?t realize there was a policy about load testing > questions. What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I > do I will > > provide you with the feedback when I do. Just let me know > what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old > release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may > as well > > take the time to try it now that you have opened a forum > questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and > verify > > that each one can hear the audio perfectly and in time with > what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load > test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot > because we have > > a policy against entertaining load testing questions but if > you like > > asterisk, by all means, use it, and good luck to you if > those numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. > Are there > > substantial fixes to mod_conference in the FreeSWITCH trunk > that might > > increase capacity for my scenario of one speaker and many > listeners? > > If I want to put this into a production environment, I would > need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, > and doing > > the same scenario was able to get 1 speaker and 600 > listeners on a > > single conference with no audio issues. The CPU at that > point was just > > over 300%, same as where the single conference scenario > failed on > > FreeSWITCH with 300 listeners. I was able to push it to > over 700 > > listeners before I reached 400% CPU usage (I guess maxing > out my > > quad-core processors), and asterisk finally crashed. But up > until that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more > scalable > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more > than > > doubling FreeSWITCH capacity in this case. Again, maybe > there is > > something on the FreeSWITCH side that I?m doing wrong, but I > don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn > trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other > solutions. My > > scenario is to have one speaker, and many listeners (mute). > Since I > > have only one speaker, I was expecting this to scale well > because > > there is no audio mixing required, just send each frame of > the single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on > what I?ve > > read on how FreeSWITCH is supposed to be generally very > scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon > server, 4 Gig > > of RAM. I?ve set file logging to ?notice? level. My > conference profile > > is configured to suppress several events, hoping that it > would improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I > reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 > listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx > 110 > > listeners per conference (so just over 400 total channels on > the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the > system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH > process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar > testing, or > > maybe has suggestions on how to improve the performance. Or > perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users-list at metik.com Fri Dec 18 09:45:12 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 18 Dec 2009 12:45:12 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2B2188.2060803@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> Message-ID: <4B2BBFA8.9050900@metik.com> Honestly, several years ago I accomplished this by mod'ing SER (which became OpenSER which was then forked to OpenSIPS and Kamalio) and using one cluster of proxies for subscriber endpoints and another for infrastructure (so that I could keep RTP flows optimized yet support double NAT when required by an endpoint). Although the network looks different today. However, we were never quite happy about the lack of media failover (complicated NAT) and evaluated several commercial solutions until finding Covergence (which is now, for better or for worse since the jury is still out, owned by ACME Packet). At the time, they offered the best mix of security (their forte) yet scaled very well in comparison to their competitors that I had tested in our lab (ACME Packet, Kagoor, Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great decision, not unlike that of the FS developers, to implement a proven/stable SIP protocol stack. Nothing is perfect and that does not mean that we did not spend a considerable amount of time documenting bugs so that they could be addressed and it would work as it should We still use OpenSIPS for certain CSCF functionality (due to its speed and flexibility since it is not a B2BUA). Based on Mathieu's response (and he is definitely someone that would know), it looks like you should be able to easily append a X-AUTH-IP header (via OpenSIPS) containing the IP address of the endpoint and call it a day. -metik Bill W wrote: > Hey Metik, > > That's exactly what I'm trying to do... load balance across multiple FS > boxes, and have any machine in the cluster be able to reach a device > behind a NAT firewall. Hence the need for the proxy. Also, I'm trying > to keep the proxy relatively "dumb" and put all the logic in the FS boxes. > > True I could do the auth on the proxies as well, but then I'm setting up > another authentication scheme in addition to what is on the FS boxes, > and then integrating the databases so everything is consistent. > > I also have hosts that talk to the FS boxes directly, rather than > through the proxy. So I can't get rid of auth_acl on FS either, even if > I do implement it on the proxies. So my setup becomes much more > complex and potentially brittle. > > And all we're really talking about for FreeSWITCH, conceptually > speaking, is populating a variable with a different IP. We could even > make it configurable, as to which IP is to be used for the auth-acl. > > What are you using for SBCs? (if you are allowed to divulge that) I'm > currently using OpenSIPS for my proxy. > > Thanks, > Bill > > Metik wrote: > >> Why not simply implement this feature in the PROXY itself? >> >> FS has a pretty comprehensive security feature set for endpoints that >> directly register with it. >> >> Don't get me wrong, I do agree this is useful especially if you are >> going to be using your proxies to load balance across multiple FS boxes >> to create an ad-hoc cluster. I actually have session border controllers >> that have this feature and use it quite often. >> >> -metik >> >> Bill W wrote: >> >>> Hey Metik, >>> >>> Thanks for the reply, and the pointers for doing it with xml_curl. >>> >>> I'll guess have to do that in the short term, but in my opinion, having >>> auth-acl be able to work through a proxy is very important as it is a >>> vital part of a comprehensive security feature set. And it would be >>> much simpler to implement from an end-user perspective than the >>> alternative of doing it in xml_curl. >>> >>> As a matter of fact, I'm considering offering a bounty for that feature. >>> What is the going rate for that kind of thing? >>> >>> Is anyone out there interested in coding this feature? Or chipping in >>> for the bounty? >>> >>> >>> Thanks, >>> Bill >>> >>> >>> Metik wrote: >>> >>> >>>> This may be difficult considering that ACL needs to consider the >>>> original src IP/URI. To do that it, freeswitch would need to do so >>>> using a header that retains that information (i.e. From, Via, Contact, >>>> etc.). Which I do not believe is currently possible using auth-acl or >>>> apply-proxy-acl. >>>> >>>> However, you should be able to emulate the behavior using mod_xml_curl >>>> (and validating against appropriate variables available when using it to >>>> authenticate the request). >>>> >>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>> >>>> -metik >>>> >>>> >>>> Bill W wrote: >>>> >>>> >>>>> Hey Brian, >>>>> >>>>> >>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>> through a proxy the way I want them to, even with setting >>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>> >>>>> I have a multi-tenant system with multiple domains with multiple users >>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>> >>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>> the proxy. >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> Brian West wrote: >>>>> >>>>> >>>>> >>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>> >>>>>> /b >>>>>> >>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Okay, I added: to my sofia >>>>>>> profile and restarted sofia, and still no joy. >>>>>>> >>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>> I've got in >>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>> >>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>> >>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>> >>>>>>> Thoughts? >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>>> >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Dec 18 09:47:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 11:47:40 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261158104.11815.92.camel@luna.tc.commsmundi.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <1261158104.11815.92.camel@luna.tc.commsmundi.com> Message-ID: <191c3a030912180947s3e027aa8g712e1639934b6fe7@mail.gmail.com> yes, I understand. My reply was to the thread in general not directed at you =p On Fri, Dec 18, 2009 at 11:41 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > It was of course just bad humor, I love both projects for what they are, > and I agree that both have their own advantages and inconvenients. > > For example, accessing that same conference from a dahdi card could be > another goal where Asterisk would be at an advantage, as chan_dahdi is > still superior (in the more tested sense) than openzap+mod_openzap. > > I just use both projects separately or together depending on what's > needed! > > I'm no banker nor do I understand the code, but many thanks for all > those unpaid contributions providing an excellent alternative for free > telephony. Your names really deserve being engraved in google's cache > for eternity. :-) > > But still, I would like to see those numbers... > > Fran?ois. > > > On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: > > Conferencing is hardly the best place to judge performance. > > Quality is a far more important goal to me in conferencing. > > > > Lets compare who can do 48khz conferences with several 32k siren > > callers on a polycom 6000, several more using G722 at 16khz and > > another handful of people on g711 ulaw all at different rates and > > ptimes talking in near-real time with low delay and low echo. The > > fact that you can broadcast the conferences to icecast, control it > > from an external application and play files etc, and oh yeah, it can > > stream video. > > > > Frankly, considering this is a free software project and so many > > people benefit, i would rather focus on quality than what numbers i > > can get from having robots call the conference in some way that > > probably does not match reality. I would love for someone to sponsor > > the effort to add features to the conference module, but of course, I > > do not hold my breath, instead I continue to improve it for free when > > I find time. This is one of many reasons I do not enjoy performance > > discussions unless I am talking to an engineer who understands the > > code or a banker ready to pay for improvements. That is not my way of > > saying pay me or forget it as you can clearly see the conference > > module has made it to where it is today with no financial support at > > all. Just the efforts of myself and several brave volunteers over the > > years who have contributed to it. > > > > BTW, > > > > We have a weekly call, there is one today in 30 minutes. > > Drop by sip:888 at conference.freeswitch.orgThis is just an openVZ > > instance mind you running at 48khz waiting for anyone to call in and > > say hi. > > > > > > > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > > wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, > > sounds like > > a configuration error. > > > > If not, I already see the title of the next Digium blog entry: > > "FreeSwitch scalability myth finally ends: The worst Asterisk > > version > > ever (1.4) beating the crap of the best and latest FS." > > > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see > > who wins > > the final conference battle! :-) > > > > Fran?ois. > > > > > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > > I did a test with the trunk version for the one conference > > case, and > > > it is the same results as for 1.0.4. The audio failed at > > around 300 > > > listeners. Oddly though, it consumed less %CPU (240% instead > > of 300%), > > > and yet the audio still failed at the same number of > > listeners. > > > > > > > > > > > > Brian. > > > > > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > > Sent: Thursday, December 17, 2009 3:49 PM > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with > > them and > > > many of them are unfounded and take up a lot of time to > > track down. > > > That does not mean you have not found a real problem but the > > first > > > step is trying trunk. > > > > > > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > > > > wrote: > > > > > > I didn?t realize there was a policy about load testing > > questions. What > > > forum should I have used for this? > > > > > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I > > do I will > > > provide you with the feedback when I do. Just let me know > > what forum > > > to use for this topic from now on. > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > One man's stable release is another man's 6 month old > > release with > > > hundreds of known fixed bugs. > > > If one of the core developers tells you to try it, you may > > as well > > > take the time to try it now that you have opened a forum > > questioning > > > the scalability. > > > > > > When you tested asterisk did you actually use 600 phones and > > verify > > > that each one can hear the audio perfectly and in time with > > what the > > > speaker was saying? Did you try same on FS? > > > > > > Did you optimize your dialplan on FS to deal with a load > > test or > > > follow any of the recommended performance tuning page. > > > > > > All of the answers to these questions are really moot > > because we have > > > a policy against entertaining load testing questions but if > > you like > > > asterisk, by all means, use it, and good luck to you if > > those numbers > > > you are testing at are what you plan to put in real > > > production......... > > > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > > > > wrote: > > > > > > Hi Mike, > > > > > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. > > Are there > > > substantial fixes to mod_conference in the FreeSWITCH trunk > > that might > > > increase capacity for my scenario of one speaker and many > > listeners? > > > If I want to put this into a production environment, I would > > need a > > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, > > and doing > > > the same scenario was able to get 1 speaker and 600 > > listeners on a > > > single conference with no audio issues. The CPU at that > > point was just > > > over 300%, same as where the single conference scenario > > failed on > > > FreeSWITCH with 300 listeners. I was able to push it to > > over 700 > > > listeners before I reached 400% CPU usage (I guess maxing > > out my > > > quad-core processors), and asterisk finally crashed. But up > > until that > > > point, there were no audio problems. > > > > > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more > > scalable > > > than Asterisk, but unless there is something wrong with my > > FreeSWITCH > > > setup, Asterisk was clearly the winner in this test ? more > > than > > > doubling FreeSWITCH capacity in this case. Again, maybe > > there is > > > something on the FreeSWITCH side that I?m doing wrong, but I > > don?t see > > > what it could be. > > > > > > > > > > > > Brian. > > > > > > > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > > Sent: Thursday, December 17, 2009 10:18 AM > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > I would be curious what the same tests produce with svn > > trunk of > > > FreeSWITCH. > > > > > > > > > > > > > > > Mike > > > > > > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > > mod_conference to see if it will scale better that other > > solutions. My > > > scenario is to have one speaker, and many listeners (mute). > > Since I > > > have only one speaker, I was expecting this to scale well > > because > > > there is no audio mixing required, just send each frame of > > the single > > > speaker to each listener. Unfortunately, my testing was > > disappointing, > > > and it didn?t scale nearly as well as I?d hoped (based on > > what I?ve > > > read on how FreeSWITCH is supposed to be generally very > > scalable). > > > > > > > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon > > server, 4 Gig > > > of RAM. I?ve set file logging to ?notice? level. My > > conference profile > > > is configured to suppress several events, hoping that it > > would improve > > > performance. > > > > > > > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I > > reached the > > > point of audio failure on the conferences: > > > > > > > > > > > > > > > > > > Scenario 1: > > > > > > > > > 1 conference, 1 speaker, audio failed at approx 300 > > listeners (mute) > > > > > > > > > > > > > > > > > > Scenario 2: > > > > > > > > > 4 conferences, 1 speaker per conference, audio failed approx > > 110 > > > listeners per conference (so just over 400 total channels on > > the > > > system). > > > > > > > > > > > > > > > > > > Scenario 3: > > > > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > > listeners > > > per conference (so just over 500 total channels on the > > system). > > > > > > > > > > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > > scenarios, > > > the audio quality failed when the % CPU for the FreeSWITCH > > process > > > exceeded 300%. > > > > > > > > > > > > > > > > > > I was hoping maybe someone else might have done similar > > testing, or > > > maybe has suggestions on how to improve the performance. Or > > perhaps an > > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > > > > > Brian. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/4394927e/attachment-0001.html From djbinter at yahoo.com Fri Dec 18 10:09:29 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 10:09:29 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <156715.67103.qm@web37503.mail.mud.yahoo.com> Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/70548a5c/attachment.html From msc at freeswitch.org Fri Dec 18 10:18:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 10:18:33 -0800 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> Message-ID: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij wrote: > Hi guys (and girls)! > > I'm working on a little bit of ENUM trickery and I tried doing some > (illegal) nested conditions. :-) > > What I want to do is to first check enum with the ENUM application, > then depending on the answer do some stuff. Say that the domain part > of the ENUM answer is robin.nl, then I want to do action X instead of > just briding the enum answer directly as I see in most examples. > > But I remembered that it wasn't allowed to do nested conditions. So > what I did was stacked conditions. After that I read the dialplan wiki > pages again and figured that my regexp never matches because variables > I "set" during some phase of the extension I can't use in the same > "go" as another condition. So, now my plan is to use LUA to do the > regexp. > > I'll pass the enum answer to a lua script which will split the answer > in a user and domain part and return those two to the main app. Then > based on those two vars I'll do routing or other actions (like, prefix > and then route). > > Is this how I'm supposed to do it? I can't find many examples on > manipulating ENUM answers, other than bridging them directly. I can't > change the way I do stuff to ENUM answers, because in most cases I'll > just route them out the standard way. > > Anyone with experience on fiddling with ENUM answers? > One thing you can do is create an extension that does the enum look up and then transfers the call back into the dialplan. You could set up a separate context that handles just the enum checking. Your condition would just need to match whatever var you put the enum return val in. So if your var name is "enum_res" then you can transfer like this after your enum lookup: Then create a context named "my_enum_context" and match for the condition(s) you need, like: do stuff Then have a different extension for other values of enum_res... This is just one way to do it without using a scripting lang. you can by all means use Lua as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/a971d7d1/attachment.html From djbinter at yahoo.com Fri Dec 18 10:23:18 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 10:23:18 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <618888.32098.qm@web37504.mail.mud.yahoo.com> Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c0704cd0/attachment-0001.html From jerry.richards at teotech.com Fri Dec 18 10:35:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 18 Dec 2009 10:35:41 -0800 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header Message-ID: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry From freeswitch-users-list at metik.com Fri Dec 18 11:13:03 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 18 Dec 2009 14:13:03 -0500 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> Message-ID: <4B2BD43F.50903@metik.com> I use a similar method (transfer to XML dialplan based on the value of "${enum_route_1}") to determine if the SIP URI is native to a particular FS instance or if it needs to be sent off-net and it works well. -metik Michael Collins wrote: > > > On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij > wrote: > > Hi guys (and girls)! > > I'm working on a little bit of ENUM trickery and I tried doing some > (illegal) nested conditions. :-) > > What I want to do is to first check enum with the ENUM application, > then depending on the answer do some stuff. Say that the domain part > of the ENUM answer is robin.nl , then I want to > do action X instead of > just briding the enum answer directly as I see in most examples. > > But I remembered that it wasn't allowed to do nested conditions. So > what I did was stacked conditions. After that I read the dialplan wiki > pages again and figured that my regexp never matches because variables > I "set" during some phase of the extension I can't use in the same > "go" as another condition. So, now my plan is to use LUA to do the > regexp. > > I'll pass the enum answer to a lua script which will split the answer > in a user and domain part and return those two to the main app. Then > based on those two vars I'll do routing or other actions (like, prefix > and then route). > > Is this how I'm supposed to do it? I can't find many examples on > manipulating ENUM answers, other than bridging them directly. I can't > change the way I do stuff to ENUM answers, because in most cases I'll > just route them out the standard way. > > Anyone with experience on fiddling with ENUM answers? > > > One thing you can do is create an extension that does the enum look up > and then transfers the call back into the dialplan. You could set up a > separate context that handles just the enum checking. Your condition > would just need to match whatever var you put the enum return val in. > So if your var name is "enum_res" then you can transfer like this > after your enum lookup: > > > > Then create a context named "my_enum_context" and match for the > condition(s) you need, like: > > do stuff > > > Then have a different extension for other values of enum_res... > > This is just one way to do it without using a scripting lang. you can > by all means use Lua as well. > -MC > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at proximosystems.com Fri Dec 18 11:14:33 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 14:14:33 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Message-ID: <00b901ca8016$55289800$ff79c800$@com> I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of ?having robots call the conference in a way that probably does not match reality?. In fact, this will very much reflect the reality of the application I?m building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum ? per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I?m trying to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:888 at conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) Fran?ois. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 3:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > We didn't post it anywhere but we just get overwhelmed with them and > many of them are unfounded and take up a lot of time to track down. > That does not mean you have not found a real problem but the first > step is trying trunk. > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/a880f3d8/attachment-0001.html From msc at freeswitch.org Fri Dec 18 11:33:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 11:33:02 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > I was evaluating the technologies available, and I thought you would be > interested in my results. However, almost every other reply I get from you > to my posts, rather than being helpful, has been hostile and insulting. > Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) > > > My scenario is not a hypothetical one of ?having robots call the conference > in a way that probably does not match reality?. In fact, this will very much > reflect the reality of the application I?m building. Only instead of 300 > listeners, I need to scale to over 2000 listeners minimum ? per event, with > possibly more than one concurrent event. I want to pack as many listeners on > one server as I can. I?m trying to find a real solution to a real problem. > That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. > > > I work with other open source projects and fund enhancements or fixes I > need. FreeSWITCH would be no different. > > > Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/792c7412/attachment.html From lon at kickasspixels.com Fri Dec 18 11:41:12 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 18 Dec 2009 11:41:12 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <2B659DC6-32F0-4046-93E5-22E3E5EEEF65@kickasspixels.com> Brian, Now that you know the scale freeswotch scales to in you scenario, and having designed a mult-server solution can you not add more server to scale further? As freeswitch continues to improve retest and revise your architecture design. Sent from my iPhone On Dec 18, 2009, at 11:14 AM, Brian wrote: > I was evaluating the technologies available, and I thought you would > be interested in my results. However, almost every other reply I get > from you to my posts, rather than being helpful, has been hostile > and insulting. > > > > My scenario is not a hypothetical one of ?having robots call the con > ference in a way that probably does not match reality?. In fact, thi > s will very much reflect the reality of the application I?m building > . Only instead of 300 listeners, I need to scale to over 2000 listen > ers minimum ? per event, with possibly more than one concurrent even > t. I want to pack as many listeners on one server as I can. I?m tryi > ng to find a real solution to a real problem. > > > > I work with other open source projects and fund enhancements or > fixes I need. FreeSWITCH would be no different. > > > > Brian. > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, December 18, 2009 11:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to > sponsor the effort to add features to the conference module, but of > course, I do not hold my breath, instead I continue to improve it > for free when I find time. This is one of many reasons I do not > enjoy performance discussions unless I am talking to an engineer who > understands the code or a banker ready to pay for improvements. > That is not my way of saying pay me or forget it as you can clearly > see the conference module has made it to where it is today with no > financial support at all. Just the efforts of myself and several > brave volunteers over the years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde m> wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds > like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. > What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I w > ill > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we > have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those > numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are th > ere > > substantial fixes to mod_conference in the FreeSWITCH trunk that > might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was > just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until > that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalab > le > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I > don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other > solutions. My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the > single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what > I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 > Gig > > of RAM. I?ve set file logging to ?notice? level. My > conference profile > > is configured to suppress several events, hoping that it would > improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or > perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/31d920c9/attachment-0001.html From viper at fx-services.com Fri Dec 18 11:51:37 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 18 Dec 2009 20:51:37 +0100 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> Message-ID: <4B2BDD49.9030908@fx-services.com> On 12/18/09 7:18 PM, Michael Collins wrote: Hi Michael, > One thing you can do is create an extension that does the enum look up > and then transfers the call back into the dialplan. You could set up a Cool idea, didn't think about that! > separate context that handles just the enum checking. Your condition > would just need to match whatever var you put the enum return val in. So > if your var name is "enum_res" then you can transfer like this after > your enum lookup: Right, makes sense. Going to try a bit in that direction. Do the enum lookup and then transfer to an enum handling context. Simple, should have thought about that. :) > This is just one way to do it without using a scripting lang. you can by > all means use Lua as well. My main question there really was, since I'm not able to work on vars I set in an extention, will that work if I return vars from a script? It should really, but I was asking to make sure it would. I think in that design, the script would have been like three rules or something, but keeping it in the dialplan is nicer, I think (even though it says "don't do magic in the dialplan, do it in scripts" on the wiki). I'll report back when I managed to fiddle something together. /Robin From anthony.minessale at gmail.com Fri Dec 18 11:55:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 13:55:38 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> Message-ID: <191c3a030912181155h411877c9y9b9d40849f54dd0@mail.gmail.com> Brian, there was not one insulting word in anything I have said and as this is a community mailing list my replies are always voiced to address the public in general not you specifically, like I already mentioned in my last post. If you open a public forum on a FAQ be prepared to hear our policy. Indeed many people do unrealistic load testing and most people with strong will find it insulting when a group of people have a set of standard policy by which they try to deal with making a penny jar for all the 2 cents worth of input we get on a daily basis. I can't begin to iterate over all the cases we endure on a weekly basis. additionally 90% of bug reports are on older releases and we always make people reproduce their issues on SVN trunk because 3 core devs and a handful of helpers can't maintain 20 versions of the code. I gave you some really suggestions yesterday let me repaste it, I fail to see any insults: --------------------------------------------------------------------------------------------------------------------------------------------------- What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp --------------------------------------------------------------------------------------------------------------------------------------------------- I have to get in these "fights" with people constantly so I guess that is part of my job and my biggest mistake is spending so much time trying to explain myself. - Show quoted text - On Fri, Dec 18, 2009 at 1:33 PM, Michael Collins wrote: > > > On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > >> I was evaluating the technologies available, and I thought you would be >> interested in my results. However, almost every other reply I get from you >> to my posts, rather than being helpful, has been hostile and insulting. >> > Thanks for your input. Just so you know, Tony deals with people on a near > daily basis who want to spend time doing crazy schemes under the guise of > "load testing" or "researching a new solution" which are not grounded in > reality. At first blush this scenario sounded like one of those schemes. > However it definitely looks like you've built a test scenario that mimics > reality better than most. I think we can give you a pass for not being able > to get 500 people all at once to call in every time you need to test. :) > >> >> >> My scenario is not a hypothetical one of ?having robots call the >> conference in a way that probably does not match reality?. In fact, this >> will very much reflect the reality of the application I?m building. Only >> instead of 300 listeners, I need to scale to over 2000 listeners minimum ? >> per event, with possibly more than one concurrent event. I want to pack as >> many listeners on one server as I can. I?m trying to find a real solution to >> a real problem. >> > That kind of volume suggests that the icecast style solution would be best. > It takes much less resources to send audio in one direction than it does to > mix audio from multiple parties. I like bkw's initial suggestion of > transferring a caller to the conference only when he/she needs to speak, > such as to ask a question. Like Tony mentioned, his focus is on quality not > quantity, so mod_conference probably isn't the best tool for this scenario. > >> >> >> I work with other open source projects and fund enhancements or fixes I >> need. FreeSWITCH would be no different. >> >> >> > Excellent! It looks like we don't already have a canned solution, > obviously, but as bkw likes to say, all the Lego bricks are there to build > the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the > weekly conference which is going on right now and you might catch some of > the devs and leading community members and you can chat in real-time about > your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) > > -Michael > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/35731e70/attachment.html From anthony.minessale at gmail.com Fri Dec 18 12:02:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:02:14 -0600 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header In-Reply-To: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> References: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Message-ID: <191c3a030912181202h471fa07n3f974664b64b6b1b@mail.gmail.com> could be possible with a code change, open a bounty on jira and someone may do it On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards wrote: > Is it possible to allow/deny REGISTER requests based on the User-Agent > header? I need to know/manage what devices are registering. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/38fb823a/attachment.html From yehavi.bourvine at gmail.com Fri Dec 18 12:16:04 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 18 Dec 2009 22:16:04 +0200 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <26842055.post@talk.nabble.com> References: <26842055.post@talk.nabble.com> Message-ID: Try the following: I don't know whether it will work in your case, but here we use it to reject a call while we want to signal that the remote party is busy. Regards, __Yehavi: 2009/12/18 bcxml > > I have an incomming call being answered by FreeSwitch and passed to IVR > application which rejects the call. > > The call is never answered by FreeSwitch, but instead of hearing a busy > signal, the caller hears ringing. > > Can anyone advise how I can get the user to hear a busy signal after call > rejection instead of ringing. > > Here is the debug trace > > http://pastebin.freeswitch.org/11558 > > Thanks > > > Brian > > -- > View this message in context: > http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/4eb2b2ff/attachment.html From brian at proximosystems.com Fri Dec 18 12:16:28 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 15:16:28 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> Message-ID: <00d801ca801e$fb7d5670$f2780350$@com> Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/5e61f879/attachment-0001.html From anthony.minessale at gmail.com Fri Dec 18 12:30:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:30:29 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00d801ca801e$fb7d5670$f2780350$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> <00d801ca801e$fb7d5670$f2780350$@com> Message-ID: <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consulting at freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian wrote: > Hi Michael, > > > > Thanks for the invite, but I can?t make it on the call. Anyway, I?m not > sure if discussing my specific case is meant for that type of call, is it? > > > > After Brian?s suggestion to use shoutcast and local streams, I was looking > at the code for those modules. I?m not familiar with shoutcast or icecast > capabilities, so I don?t know if they can just pass though my audio stream > unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on > the source server, and then back from mp3 to uLaw (or whatever phone codec) > on the other server. > > > > I was wondering if maybe there was a way to make a stream out of an > existing channel, and have all the other channels just listen to that > stream. It would be sort of halfway between conference and shoutcast. I > would call in to the secondary server like I already do, but only instead of > entering into a conference as a speaker, the channel would just start > producing a local audio stream for the listener channels to tap into. It > would avoid the need to have another piece of software to manage (shoutcast > or icecast), and my support team would be happier... > > > > However, I would still need to do tests for the streaming idea to see how > that scales... > > > > Brian. > > > > > > *From:* Michael Collins [mailto:msc at freeswitch.org] > *Sent:* Friday, December 18, 2009 2:33 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > > > On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > > I was evaluating the technologies available, and I thought you would be > interested in my results. However, almost every other reply I get from you > to my posts, rather than being helpful, has been hostile and insulting. > > Thanks for your input. Just so you know, Tony deals with people on a near > daily basis who want to spend time doing crazy schemes under the guise of > "load testing" or "researching a new solution" which are not grounded in > reality. At first blush this scenario sounded like one of those schemes. > However it definitely looks like you've built a test scenario that mimics > reality better than most. I think we can give you a pass for not being able > to get 500 people all at once to call in every time you need to test. :) > > > > My scenario is not a hypothetical one of ?having robots call the conference > in a way that probably does not match reality?. In fact, this will very much > reflect the reality of the application I?m building. Only instead of 300 > listeners, I need to scale to over 2000 listeners minimum ? per event, with > possibly more than one concurrent event. I want to pack as many listeners on > one server as I can. I?m trying to find a real solution to a real problem. > > That kind of volume suggests that the icecast style solution would be > best. It takes much less resources to send audio in one direction than it > does to mix audio from multiple parties. I like bkw's initial suggestion of > transferring a caller to the conference only when he/she needs to speak, > such as to ask a question. Like Tony mentioned, his focus is on quality not > quantity, so mod_conference probably isn't the best tool for this scenario. > > > > I work with other open source projects and fund enhancements or fixes I > need. FreeSWITCH would be no different. > > > > Excellent! It looks like we don't already have a canned solution, > obviously, but as bkw likes to say, all the Lego bricks are there to build > the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the > weekly conference which is going on right now and you might catch some of > the devs and leading community members and you can chat in real-time about > your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) > > -Michael > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/84eb348c/attachment.html From anthony.minessale at gmail.com Fri Dec 18 12:31:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:31:28 -0600 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: References: <26842055.post@talk.nabble.com> Message-ID: <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> that will only work if you have not answered yet. if you already have, you would need to indicate the tones inband like I mentioned. On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine wrote: > Try the following: > > > I don't know whether it will work in your case, but here we use it to > reject a call while we want to signal that the remote party is busy. > > Regards, __Yehavi: > > > > 2009/12/18 bcxml > > >> I have an incomming call being answered by FreeSwitch and passed to IVR >> application which rejects the call. >> >> The call is never answered by FreeSwitch, but instead of hearing a busy >> signal, the caller hears ringing. >> >> Can anyone advise how I can get the user to hear a busy signal after call >> rejection instead of ringing. >> >> Here is the debug trace >> >> http://pastebin.freeswitch.org/11558 >> >> Thanks >> >> >> Brian >> >> -- >> View this message in context: >> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/5afd7076/attachment.html From dave at 3c.co.uk Fri Dec 18 12:56:41 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 18 Dec 2009 13:56:41 -0700 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <1261169801.6033.3.camel@local.freepabx.com> Hi Brian, Have a look at this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop - I took a quick look through the code and couldn't see any reason why you shouldn't have a bunch of eavesdroppers listening to a single caller. I'd be surprised if this didn't perform a lot better for your application. Cheers -- Dave > I was evaluating the technologies available, and I thought you would > be interested in my results. However, almost every other reply I get > from you to my posts, rather than being helpful, has been hostile and > insulting. > > > > My scenario is not a hypothetical one of ?having robots call the > conference in a way that probably does not match reality?. In fact, > this will very much reflect the reality of the application I?m > building. Only instead of 300 listeners, I need to scale to over 2000 > listeners minimum ? per event, with possibly more than one concurrent > event. I want to pack as many listeners on one server as I can. I?m > trying to find a real solution to a real problem. > > > > I work with other open source projects and fund enhancements or fixes > I need. FreeSWITCH would be no different. > > > > Brian. > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, December 18, 2009 11:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to sponsor > the effort to add features to the conference module, but of course, I > do not hold my breath, instead I continue to improve it for free when > I find time. This is one of many reasons I do not enjoy performance > discussions unless I am talking to an engineer who understands the > code or a banker ready to pay for improvements. That is not my way of > saying pay me or forget it as you can clearly see the conference > module has made it to where it is today with no financial support at > all. Just the efforts of myself and several brave volunteers over the > years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds > like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. > What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I > will > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we > have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those > numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are > there > > substantial fixes to mod_conference in the FreeSWITCH trunk that > might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was > just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until > that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I don?t > see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other solutions. > My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the > single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 > Gig > > of RAM. I?ve set file logging to ?notice? level. My conference > profile > > is configured to suppress several events, hoping that it would > improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or perhaps > an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From djbinter at yahoo.com Fri Dec 18 13:09:54 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 13:09:54 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <707033.51380.qm@web37501.mail.mud.yahoo.com> Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that I thought it was a bug. Thank you again for the community and your support. Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/dfcb75e4/attachment.html From brian at proximosystems.com Fri Dec 18 13:15:53 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 16:15:53 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> <00d801ca801e$fb7d5670$f2780350$@com> <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> Message-ID: <010c01ca8027$48521820$d8f64860$@com> Sounds like a plan. We will pursue it through the consulting at freeswith.org route. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, December 18, 2009 3:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consulting at freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian wrote: Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/335fdf8c/attachment-0001.html From bcxml at hotmail.com Fri Dec 18 14:16:56 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 18 Dec 2009 14:16:56 -0800 (PST) Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> References: <26842055.post@talk.nabble.com> <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> Message-ID: <26850453.post@talk.nabble.com> Actually my application returns 403 if it decides that it doesnt want to answer the call So I changed the response that my applicatgion gives to 486 and I now get the behavior that I wanted Thanks for the advice Brian Anthony Minessale-2 wrote: > > that will only work if you have not answered yet. > if you already have, you would need to indicate the tones inband like I > mentioned. > > > On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine > wrote: > >> Try the following: >> >> >> I don't know whether it will work in your case, but here we use it to >> reject a call while we want to signal that the remote party is busy. >> >> Regards, __Yehavi: >> >> >> >> 2009/12/18 bcxml >> >> >>> I have an incomming call being answered by FreeSwitch and passed to IVR >>> application which rejects the call. >>> >>> The call is never answered by FreeSwitch, but instead of hearing a busy >>> signal, the caller hears ringing. >>> >>> Can anyone advise how I can get the user to hear a busy signal after >>> call >>> rejection instead of ringing. >>> >>> Here is the debug trace >>> >>> http://pastebin.freeswitch.org/11558 >>> >>> Thanks >>> >>> >>> Brian >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26850453.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From frank at carmickle.com Fri Dec 18 14:22:45 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 18 Dec 2009 17:22:45 -0500 Subject: [Freeswitch-users] packaging preference question Message-ID: <20091218222245.GE31924@base.carmickle.com> Hello all The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? --FC From quentusrex at gmail.com Fri Dec 18 14:40:32 2009 From: quentusrex at gmail.com (William King) Date: Fri, 18 Dec 2009 14:40:32 -0800 Subject: [Freeswitch-users] packaging preference question In-Reply-To: <20091218222245.GE31924@base.carmickle.com> References: <20091218222245.GE31924@base.carmickle.com> Message-ID: <1261176032.8965.14.camel@quentusrex-desktop> I think that sounds like a good idea. It would also keep permission management simple. -William King On Fri, 2009-12-18 at 17:22 -0500, Frank Carmickle wrote: > Hello all > > The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at hijacked.us Fri Dec 18 17:43:59 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 18 Dec 2009 20:43:59 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) Message-ID: <20091219014359.GA21798@hijacked.us> So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version >= R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew From andrew at hijacked.us Fri Dec 18 19:16:49 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 18 Dec 2009 22:16:49 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091219014359.GA21798@hijacked.us> References: <20091219014359.GA21798@hijacked.us> Message-ID: <20091219031649.GA1956@hijacked.us> I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our own CRM - spicecsm. The URL supports interpolation for variables like callerid, clientid, call type, etc. The supervisor view is a little hard to describe via static images, but you're able to drag and drop agents into another profile (empty profiles are hidden when not dragging an agent), drag agents onto an agent to send them the call, and there's also various right click menus available. Oh, and I forgot to mention this before; this system is in 'live testing' and the goal is to do a final deployment sometime in January. Andrew From freeswitch at aastral.net Fri Dec 18 21:37:24 2009 From: freeswitch at aastral.net (Bill W.) Date: Sat, 19 Dec 2009 00:37:24 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2BBFA8.9050900@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> Message-ID: <4B2C6694.3060400@aastral.net> Hey Metik, Thanks so much for your insights and your help. And yes, I was able to append the X-AUTH-IP header with no problem. But that didn't solve the issue. After some more research, it appears that the problem isn't with auth-calls at all. I disabled all auth-call directives in every sip profile and the registration through a proxy is still being rejected. I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the ip variable against the auth_acl cidr. if (auth_acl) { if (!switch_check_network_list_ip(ip, auth_acl)) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); ret = AUTH_FORBIDDEN; goto end; } So I guess the question is, is it possible to control what gets put into the ip variable? Thanks, Bill Metik wrote: > Honestly, several years ago I accomplished this by mod'ing SER (which > became OpenSER which was then forked to OpenSIPS and Kamalio) and using > one cluster of proxies for subscriber endpoints and another for > infrastructure (so that I could keep RTP flows optimized yet support > double NAT when required by an endpoint). Although the network looks > different today. > > However, we were never quite happy about the lack of media failover > (complicated NAT) and evaluated several commercial solutions until > finding Covergence (which is now, for better or for worse since the jury > is still out, owned by ACME Packet). At the time, they offered the best > mix of security (their forte) yet scaled very well in comparison to > their competitors that I had tested in our lab (ACME Packet, Kagoor, > Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great > decision, not unlike that of the FS developers, to implement a > proven/stable SIP protocol stack. Nothing is perfect and that does not > mean that we did not spend a considerable amount of time documenting > bugs so that they could be addressed and it would work as it should > > We still use OpenSIPS for certain CSCF functionality (due to its speed > and flexibility since it is not a B2BUA). > > Based on Mathieu's response (and he is definitely someone that would > know), it looks like you should be able to easily append a X-AUTH-IP > header (via OpenSIPS) containing the IP address of the endpoint and call > it a day. > > -metik > > > Bill W wrote: >> Hey Metik, >> >> That's exactly what I'm trying to do... load balance across multiple FS >> boxes, and have any machine in the cluster be able to reach a device >> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >> >> True I could do the auth on the proxies as well, but then I'm setting up >> another authentication scheme in addition to what is on the FS boxes, >> and then integrating the databases so everything is consistent. >> >> I also have hosts that talk to the FS boxes directly, rather than >> through the proxy. So I can't get rid of auth_acl on FS either, even if >> I do implement it on the proxies. So my setup becomes much more >> complex and potentially brittle. >> >> And all we're really talking about for FreeSWITCH, conceptually >> speaking, is populating a variable with a different IP. We could even >> make it configurable, as to which IP is to be used for the auth-acl. >> >> What are you using for SBCs? (if you are allowed to divulge that) I'm >> currently using OpenSIPS for my proxy. >> >> Thanks, >> Bill >> >> Metik wrote: >> >>> Why not simply implement this feature in the PROXY itself? >>> >>> FS has a pretty comprehensive security feature set for endpoints that >>> directly register with it. >>> >>> Don't get me wrong, I do agree this is useful especially if you are >>> going to be using your proxies to load balance across multiple FS boxes >>> to create an ad-hoc cluster. I actually have session border controllers >>> that have this feature and use it quite often. >>> >>> -metik >>> >>> Bill W wrote: >>> >>>> Hey Metik, >>>> >>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>> >>>> I'll guess have to do that in the short term, but in my opinion, having >>>> auth-acl be able to work through a proxy is very important as it is a >>>> vital part of a comprehensive security feature set. And it would be >>>> much simpler to implement from an end-user perspective than the >>>> alternative of doing it in xml_curl. >>>> >>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>> What is the going rate for that kind of thing? >>>> >>>> Is anyone out there interested in coding this feature? Or chipping in >>>> for the bounty? >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> >>>> Metik wrote: >>>> >>>> >>>>> This may be difficult considering that ACL needs to consider the >>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>> apply-proxy-acl. >>>>> >>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>> (and validating against appropriate variables available when using it to >>>>> authenticate the request). >>>>> >>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>> >>>>> -metik >>>>> >>>>> >>>>> Bill W wrote: >>>>> >>>>> >>>>>> Hey Brian, >>>>>> >>>>>> >>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>> through a proxy the way I want them to, even with setting >>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>> >>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>> >>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>> the proxy. >>>>>> >>>>>> >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> Brian West wrote: >>>>>> >>>>>> >>>>>> >>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Okay, I added: to my sofia >>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>> >>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>> I've got in >>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>> >>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>> >>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>> >>>>>>>> Thoughts? >>>>>>>> Thanks, >>>>>>>> Bill >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ron.freeswitch at mcleodnet.com Fri Dec 18 21:42:36 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Fri, 18 Dec 2009 21:42:36 -0800 Subject: [Freeswitch-users] Park with Pre Answer Message-ID: <6247DAB4ECC5499180AE946F843D5C09@fromage> Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/bb71c48e/attachment-0001.html From jason at jasonjgw.net Sat Dec 19 00:31:42 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 19 Dec 2009 19:31:42 +1100 Subject: [Freeswitch-users] RTP problems in recent revisions? Message-ID: <20091219083142.GA21558@jdc.jasonjgw.net> Revision 15904 is fine, but after upgrading to revision 16003 I get the following. 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). 2. A PCMU call to a SIP provider is fine for the first 20 to 30 seconds, then the audio breaks up completely. I have ZRTP compiled in, if that makes any difference. Obviously there's a regression somewhere. Let me know if I can provide further help. From mike at jerris.com Sat Dec 19 06:25:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Dec 2009 09:25:22 -0500 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <20091219083142.GA21558@jdc.jasonjgw.net> References: <20091219083142.GA21558@jdc.jasonjgw.net> Message-ID: <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Sat Dec 19 06:25:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Dec 2009 09:25:22 -0500 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <20091219083142.GA21558@jdc.jasonjgw.net> References: <20091219083142.GA21558@jdc.jasonjgw.net> Message-ID: <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Sat Dec 19 07:19:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 09:19:54 -0600 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> References: <20091219083142.GA21558@jdc.jasonjgw.net> <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> Message-ID: <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> Also retest with no zrtp send a full console debug log with sip trace On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/5b11a82b/attachment.html From anthony.minessale at gmail.com Sat Dec 19 08:27:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 10:27:21 -0600 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <6247DAB4ECC5499180AE946F843D5C09@fromage> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> Message-ID: <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> how are you parking it? do you have a debug log showing it happen? On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod wrote: > Is there any way to park a channel without causing pre-answer (resulting > is a SIP 183 Session Progress)? > > > > Thanks, > > Ron > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/3e841f1c/attachment.html From freeswitch-users-list at metik.com Sat Dec 19 08:41:06 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sat, 19 Dec 2009 11:41:06 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2C6694.3060400@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> Message-ID: <4B2D0222.7060609@metik.com> Bill, I think you would add this to the user profile in the directory. The "brian.xml" example (located in ${confdir}/directory/) provided with the default/sample configuration files demonstrates how to to do this by introducing a "cidr" attribute to the the "user" element. Example: "http://wiki.freeswitch.org/wiki/Acl" contains some great info (including a relevant example). -metik Bill W. wrote: > Hey Metik, > > Thanks so much for your insights and your help. And yes, I was able to > append the X-AUTH-IP header with no problem. But that didn't solve the > issue. After some more research, it appears that the problem isn't with > auth-calls at all. > > I disabled all auth-call directives in every sip profile and the > registration through a proxy is still being rejected. > > I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the > ip variable against the auth_acl cidr. > > if (auth_acl) { > if (!switch_check_network_list_ip(ip, auth_acl)) { > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); > ret = AUTH_FORBIDDEN; > goto end; > } > > So I guess the question is, is it possible to control what gets put into > the ip variable? > > Thanks, > Bill > > > Metik wrote: > >> Honestly, several years ago I accomplished this by mod'ing SER (which >> became OpenSER which was then forked to OpenSIPS and Kamalio) and using >> one cluster of proxies for subscriber endpoints and another for >> infrastructure (so that I could keep RTP flows optimized yet support >> double NAT when required by an endpoint). Although the network looks >> different today. >> >> However, we were never quite happy about the lack of media failover >> (complicated NAT) and evaluated several commercial solutions until >> finding Covergence (which is now, for better or for worse since the jury >> is still out, owned by ACME Packet). At the time, they offered the best >> mix of security (their forte) yet scaled very well in comparison to >> their competitors that I had tested in our lab (ACME Packet, Kagoor, >> Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great >> decision, not unlike that of the FS developers, to implement a >> proven/stable SIP protocol stack. Nothing is perfect and that does not >> mean that we did not spend a considerable amount of time documenting >> bugs so that they could be addressed and it would work as it should >> >> We still use OpenSIPS for certain CSCF functionality (due to its speed >> and flexibility since it is not a B2BUA). >> >> Based on Mathieu's response (and he is definitely someone that would >> know), it looks like you should be able to easily append a X-AUTH-IP >> header (via OpenSIPS) containing the IP address of the endpoint and call >> it a day. >> >> -metik >> >> >> Bill W wrote: >> >>> Hey Metik, >>> >>> That's exactly what I'm trying to do... load balance across multiple FS >>> boxes, and have any machine in the cluster be able to reach a device >>> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >>> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >>> >>> True I could do the auth on the proxies as well, but then I'm setting up >>> another authentication scheme in addition to what is on the FS boxes, >>> and then integrating the databases so everything is consistent. >>> >>> I also have hosts that talk to the FS boxes directly, rather than >>> through the proxy. So I can't get rid of auth_acl on FS either, even if >>> I do implement it on the proxies. So my setup becomes much more >>> complex and potentially brittle. >>> >>> And all we're really talking about for FreeSWITCH, conceptually >>> speaking, is populating a variable with a different IP. We could even >>> make it configurable, as to which IP is to be used for the auth-acl. >>> >>> What are you using for SBCs? (if you are allowed to divulge that) I'm >>> currently using OpenSIPS for my proxy. >>> >>> Thanks, >>> Bill >>> >>> Metik wrote: >>> >>> >>>> Why not simply implement this feature in the PROXY itself? >>>> >>>> FS has a pretty comprehensive security feature set for endpoints that >>>> directly register with it. >>>> >>>> Don't get me wrong, I do agree this is useful especially if you are >>>> going to be using your proxies to load balance across multiple FS boxes >>>> to create an ad-hoc cluster. I actually have session border controllers >>>> that have this feature and use it quite often. >>>> >>>> -metik >>>> >>>> Bill W wrote: >>>> >>>> >>>>> Hey Metik, >>>>> >>>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>>> >>>>> I'll guess have to do that in the short term, but in my opinion, having >>>>> auth-acl be able to work through a proxy is very important as it is a >>>>> vital part of a comprehensive security feature set. And it would be >>>>> much simpler to implement from an end-user perspective than the >>>>> alternative of doing it in xml_curl. >>>>> >>>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>>> What is the going rate for that kind of thing? >>>>> >>>>> Is anyone out there interested in coding this feature? Or chipping in >>>>> for the bounty? >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> >>>>> Metik wrote: >>>>> >>>>> >>>>> >>>>>> This may be difficult considering that ACL needs to consider the >>>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>>> apply-proxy-acl. >>>>>> >>>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>>> (and validating against appropriate variables available when using it to >>>>>> authenticate the request). >>>>>> >>>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>>> >>>>>> -metik >>>>>> >>>>>> >>>>>> Bill W wrote: >>>>>> >>>>>> >>>>>> >>>>>>> Hey Brian, >>>>>>> >>>>>>> >>>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>>> through a proxy the way I want them to, even with setting >>>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>>> >>>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>>> >>>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>>> the proxy. >>>>>>> >>>>>>> >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>>> Brian West wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>>> >>>>>>>> /b >>>>>>>> >>>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> Okay, I added: to my sofia >>>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>>> >>>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>>> I've got in >>>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>>> >>>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>>> >>>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>>> >>>>>>>>> Thoughts? >>>>>>>>> Thanks, >>>>>>>>> Bill >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> ------------------------------------------------------------------------ >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Sat Dec 19 09:09:33 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 19 Dec 2009 18:09:33 +0100 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header In-Reply-To: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> References: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Message-ID: <4B2D08CD.6060408@gmx.net> we do this based XML-Curl. Jerry Richards schrieb: > Is it possible to allow/deny REGISTER requests based on the User-Agent > header? I need to know/manage what devices are registering. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch-users-list at metik.com Sat Dec 19 09:47:23 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sat, 19 Dec 2009 12:47:23 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D0222.7060609@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> Message-ID: <4B2D11AB.3080906@metik.com> I noticed a typo in my post that may easily confuse someone... should be: -metik Metik wrote: > Bill, > > I think you would add this to the user profile in the directory. The > "brian.xml" example (located in ${confdir}/directory/) provided with the > default/sample configuration files demonstrates how to to do this by > introducing a "cidr" attribute to the the "user" element. > > Example: > > > > > > > > > > > > "http://wiki.freeswitch.org/wiki/Acl" contains some great info > (including a relevant example). > > -metik > > Bill W. wrote: > >> Hey Metik, >> >> Thanks so much for your insights and your help. And yes, I was able to >> append the X-AUTH-IP header with no problem. But that didn't solve the >> issue. After some more research, it appears that the problem isn't with >> auth-calls at all. >> >> I disabled all auth-call directives in every sip profile and the >> registration through a proxy is still being rejected. >> >> I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the >> ip variable against the auth_acl cidr. >> >> if (auth_acl) { >> if (!switch_check_network_list_ip(ip, auth_acl)) { >> switch_log_printf(SWITCH_CHANNEL_LOG, >> SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); >> ret = AUTH_FORBIDDEN; >> goto end; >> } >> >> So I guess the question is, is it possible to control what gets put into >> the ip variable? >> >> Thanks, >> Bill >> >> >> Metik wrote: >> >> >>> Honestly, several years ago I accomplished this by mod'ing SER (which >>> became OpenSER which was then forked to OpenSIPS and Kamalio) and using >>> one cluster of proxies for subscriber endpoints and another for >>> infrastructure (so that I could keep RTP flows optimized yet support >>> double NAT when required by an endpoint). Although the network looks >>> different today. >>> >>> However, we were never quite happy about the lack of media failover >>> (complicated NAT) and evaluated several commercial solutions until >>> finding Covergence (which is now, for better or for worse since the jury >>> is still out, owned by ACME Packet). At the time, they offered the best >>> mix of security (their forte) yet scaled very well in comparison to >>> their competitors that I had tested in our lab (ACME Packet, Kagoor, >>> Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great >>> decision, not unlike that of the FS developers, to implement a >>> proven/stable SIP protocol stack. Nothing is perfect and that does not >>> mean that we did not spend a considerable amount of time documenting >>> bugs so that they could be addressed and it would work as it should >>> >>> We still use OpenSIPS for certain CSCF functionality (due to its speed >>> and flexibility since it is not a B2BUA). >>> >>> Based on Mathieu's response (and he is definitely someone that would >>> know), it looks like you should be able to easily append a X-AUTH-IP >>> header (via OpenSIPS) containing the IP address of the endpoint and call >>> it a day. >>> >>> -metik >>> >>> >>> Bill W wrote: >>> >>> >>>> Hey Metik, >>>> >>>> That's exactly what I'm trying to do... load balance across multiple FS >>>> boxes, and have any machine in the cluster be able to reach a device >>>> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >>>> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >>>> >>>> True I could do the auth on the proxies as well, but then I'm setting up >>>> another authentication scheme in addition to what is on the FS boxes, >>>> and then integrating the databases so everything is consistent. >>>> >>>> I also have hosts that talk to the FS boxes directly, rather than >>>> through the proxy. So I can't get rid of auth_acl on FS either, even if >>>> I do implement it on the proxies. So my setup becomes much more >>>> complex and potentially brittle. >>>> >>>> And all we're really talking about for FreeSWITCH, conceptually >>>> speaking, is populating a variable with a different IP. We could even >>>> make it configurable, as to which IP is to be used for the auth-acl. >>>> >>>> What are you using for SBCs? (if you are allowed to divulge that) I'm >>>> currently using OpenSIPS for my proxy. >>>> >>>> Thanks, >>>> Bill >>>> >>>> Metik wrote: >>>> >>>> >>>> >>>>> Why not simply implement this feature in the PROXY itself? >>>>> >>>>> FS has a pretty comprehensive security feature set for endpoints that >>>>> directly register with it. >>>>> >>>>> Don't get me wrong, I do agree this is useful especially if you are >>>>> going to be using your proxies to load balance across multiple FS boxes >>>>> to create an ad-hoc cluster. I actually have session border controllers >>>>> that have this feature and use it quite often. >>>>> >>>>> -metik >>>>> >>>>> Bill W wrote: >>>>> >>>>> >>>>> >>>>>> Hey Metik, >>>>>> >>>>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>>>> >>>>>> I'll guess have to do that in the short term, but in my opinion, having >>>>>> auth-acl be able to work through a proxy is very important as it is a >>>>>> vital part of a comprehensive security feature set. And it would be >>>>>> much simpler to implement from an end-user perspective than the >>>>>> alternative of doing it in xml_curl. >>>>>> >>>>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>>>> What is the going rate for that kind of thing? >>>>>> >>>>>> Is anyone out there interested in coding this feature? Or chipping in >>>>>> for the bounty? >>>>>> >>>>>> >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> >>>>>> Metik wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> This may be difficult considering that ACL needs to consider the >>>>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>>>> apply-proxy-acl. >>>>>>> >>>>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>>>> (and validating against appropriate variables available when using it to >>>>>>> authenticate the request). >>>>>>> >>>>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>>>> >>>>>>> -metik >>>>>>> >>>>>>> >>>>>>> Bill W wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Hey Brian, >>>>>>>> >>>>>>>> >>>>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>>>> through a proxy the way I want them to, even with setting >>>>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>>>> >>>>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>>>> >>>>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>>>> the proxy. >>>>>>>> >>>>>>>> >>>>>>>> Thanks, >>>>>>>> Bill >>>>>>>> >>>>>>>> Brian West wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>>>> >>>>>>>>> /b >>>>>>>>> >>>>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>> Okay, I added: to my sofia >>>>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>>>> >>>>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>>>> I've got in >>>>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>>>> >>>>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>>>> >>>>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>>>> >>>>>>>>>> Thoughts? >>>>>>>>>> Thanks, >>>>>>>>>> Bill >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> ------------------------------------------------------------------------ >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sat Dec 19 10:29:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 12:29:10 -0600 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> References: <20091219083142.GA21558@jdc.jasonjgw.net> <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> Message-ID: <191c3a030912191029u5637df13v65509e42743af22d@mail.gmail.com> I tried a patch out of pure deduction and speculation from your post. Can you update and test it for me please? On Sat, Dec 19, 2009 at 9:19 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Also retest with no zrtp > send a full console debug log with sip trace > > On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: > > The best help to track this down is to try to identify the specific > svn revision that caused the issue and to supply a full freeswitch > debug with sip trace. > > Mike > > On Dec 19, 2009, at 3:31 AM, Jason White wrote: > > Revision 15904 is fine, but... > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/bfc974aa/attachment.html From ron.freeswitch at mcleodnet.com Sat Dec 19 10:54:56 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 10:54:56 -0800 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> Message-ID: <347A9A7174A14557B5ADC2900B21A784@fromage> This is what I am doing . DIALPLAN NETWORK TRACE 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE sip:6042772011 at 192.168.100.132:5090, with session description 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy Authentication Required 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK sip:6042772011 at 192.168.100.132:5090 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE sip:6042772011 at 192.168.100.132:5090, with session description 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 Session Progress, with session description CONSOLE 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected by acl "domains". Falling back to Digest auth. 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected by acl "domains". Falling back to Digest auth. 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel sofia/internal/695 at 192.168.100.132:5060 [07a14700-eccf-11de-8080-6fed700309ce] 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: v=0 o=695 123456 654323 IN IP4 192.168.100.140 s=none c=IN IP4 192.168.100.140 t=0 0 m=audio 10900 RTP/AVP 0 8 18 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/695 at 192.168.100.132:5060) State NEW 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/695 at 192.168.100.132:5060) State INIT 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> CS_ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/695 at 192.168.100.132:5060) State ROUTING 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing Phone 300->6042772011 in context mytest Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest->mytest] continue=false Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] destination_number(6042772011) =~ /^.*$/ break=on-false Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> CS_EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 Application park Requires media! pre_answering channel sofia/internal/695 at 192.168.100.132:5060 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending early media 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> 192.168.100.140 port 10900 codec: 0 ms: 20 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 s=FreeSWITCH c=IN IP4 192.168.100.132 t=0 0 m=audio 25382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/695 at 192.168.100.132:5060! 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, December 19, 2009 8:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Park with Pre Answer how are you parking it? do you have a debug log showing it happen? On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod wrote: Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/5ac5c7f1/attachment-0001.html From frank at carmickle.com Sat Dec 19 12:19:43 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sat, 19 Dec 2009 15:19:43 -0500 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <347A9A7174A14557B5ADC2900B21A784@fromage> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> <347A9A7174A14557B5ADC2900B21A784@fromage> Message-ID: <20091219201943.GG31924@base.carmickle.com> Good afternoon By default the internal profile is looking to have authed calls. If you want you can set an acl. Look at autoload_configs/acl.conf.xml. Also remember to set the context in the profile so that the dialplan for that context will be parsed. If you decide to register to it you can set the context in the directory entry for that user. Let us know how you do. --FC On Sat, Dec 19, Ron McLeod wrote: > This is what I am doing . > > > > DIALPLAN > > > > > > > > > > > > > > > > > > > > > > > > > > NETWORK TRACE > > 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > sip:6042772011 at 192.168.100.132:5090, with session description > > 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy > Authentication Required > > 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK > sip:6042772011 at 192.168.100.132:5090 > > 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > sip:6042772011 at 192.168.100.132:5090, with session description > > 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 Session > Progress, with session description > > > > > > CONSOLE > > 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/695 at 192.168.100.132:5060 > [07a14700-eccf-11de-8080-6fed700309ce] > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel > sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: > > v=0 > > o=695 123456 654323 IN IP4 192.168.100.140 > > s=none > > c=IN IP4 192.168.100.140 > > t=0 0 > > m=audio 10900 RTP/AVP 0 8 18 2 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729A/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:2 G726-32/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 > (sofia/internal/695 at 192.168.100.132:5060) State NEW > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec > sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload > to 101 > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/695 at 192.168.100.132:5060) State INIT > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 > sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> CS_ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 > sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING > > 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing Phone > 300->6042772011 in context mytest > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest->mytest] > continue=false > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] > destination_number(6042772011) =~ /^.*$/ break=on-false > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> > CS_EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 > sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 > sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 Application > park Requires media! pre_answering channel > sofia/internal/695 at 192.168.100.132:5060 > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 > sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] > > 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending early > media > > 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP > [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> > 192.168.100.140 port 10900 codec: 0 ms: 20 > > 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer [soft] > 160 bytes per 20ms > > 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: > > v=0 > > o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 > > s=FreeSWITCH > > c=IN IP4 192.168.100.132 > > t=0 0 > > m=audio 25382 RTP/AVP 0 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/695 at 192.168.100.132:5060! > > 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() > > 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel > sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] > > > > > > > > _____ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Saturday, December 19, 2009 8:27 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Park with Pre Answer > > > > how are you parking it? > do you have a debug log showing it happen? > > > > On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod > wrote: > > Is there any way to park a channel without causing pre-answer (resulting is > a SIP 183 Session Progress)? > > > > Thanks, > > Ron > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ron.freeswitch at mcleodnet.com Sat Dec 19 12:35:39 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 12:35:39 -0800 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <20091219201943.GG31924@base.carmickle.com> References: <6247DAB4ECC5499180AE946F843D5C09@fromage><191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com><347A9A7174A14557B5ADC2900B21A784@fromage> <20091219201943.GG31924@base.carmickle.com> Message-ID: <5872485EBB7443D1BFBC60D51CAF5578@fromage> My issue has nothing to do with registration or authentication. I am simply looking for a way to park a new call without having the call "pre-answered" (I don't want a SIP 183 sent back to the client). Ron > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Frank Carmickle > Sent: Saturday, December 19, 2009 12:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Park with Pre Answer > > Good afternoon > > By default the internal profile is looking to have authed calls. If you > want you can set an acl. Look at autoload_configs/acl.conf.xml. > > Also remember to set the context in the profile so that the dialplan for > that context will be parsed. If you decide to register to it you can set > the context in the directory entry for that user. > > Let us know how you do. > --FC > > > On Sat, Dec 19, Ron McLeod wrote: > > This is what I am doing . > > > > > > > > DIALPLAN > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > NETWORK TRACE > > > > 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > > sip:6042772011 at 192.168.100.132:5090, with session description > > > > 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > > > 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy > > Authentication Required > > > > 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK > > sip:6042772011 at 192.168.100.132:5090 > > > > 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > > sip:6042772011 at 192.168.100.132:5090, with session description > > > > 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > > > 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 > Session > > Progress, with session description > > > > > > > > > > > > CONSOLE > > > > 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 > Rejected > > by acl "domains". Falling back to Digest auth. > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 > Rejected > > by acl "domains". Falling back to Digest auth. > > > > 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel > > sofia/internal/695 at 192.168.100.132:5060 > > [07a14700-eccf-11de-8080-6fed700309ce] > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel > > sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: > > > > v=0 > > > > o=695 123456 654323 IN IP4 192.168.100.140 > > > > s=none > > > > c=IN IP4 192.168.100.140 > > > > t=0 0 > > > > m=audio 10900 RTP/AVP 0 8 18 2 101 > > > > a=rtpmap:0 PCMU/8000 > > > > a=rtpmap:8 PCMA/8000 > > > > a=rtpmap:18 G729A/8000 > > > > a=fmtp:18 annexb=no > > > > a=rtpmap:2 G726-32/8000 > > > > a=rtpmap:101 telephone-event/8000 > > > > a=fmtp:101 0-15 > > > > a=ptime:20 > > > > > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 > > (sofia/internal/695 at 192.168.100.132:5060) State NEW > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G7221:115:32000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G7221:107:16000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G722:9:8000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[PCMU:0:8000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec > > sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf > payload > > to 101 > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > > (sofia/internal/695 at 192.168.100.132:5060) State INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> > CS_ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > > (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change > CS_ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 > > sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING > > > > 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing > Phone > > 300->6042772011 in context mytest > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest- > >mytest] > > continue=false > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] > > destination_number(6042772011) =~ /^.*$/ break=on-false > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> > > CS_EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change > CS_EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 > > (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 > > sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 > Application > > park Requires media! pre_answering channel > > sofia/internal/695 at 192.168.100.132:5060 > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 > > sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] > > > > 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending > early > > media > > > > 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP > > [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> > > 192.168.100.140 port 10900 codec: 0 ms: 20 > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer > [soft] > > 160 bytes per 20ms > > > > 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: > > > > v=0 > > > > o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 > > > > s=FreeSWITCH > > > > c=IN IP4 192.168.100.132 > > > > t=0 0 > > > > m=audio 25382 RTP/AVP 0 101 > > > > a=rtpmap:0 PCMU/8000 > > > > a=rtpmap:101 telephone-event/8000 > > > > a=fmtp:101 0-16 > > > > a=silenceSupp:off - - - - > > > > a=ptime:20 > > > > a=sendrecv > > > > > > > > 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer > > sofia/internal/695 at 192.168.100.132:5060! > > > > 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() > > > > 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel > > sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] > > > > > > > > > > > > > > > > _____ > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: Saturday, December 19, 2009 8:27 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Park with Pre Answer > > > > > > > > how are you parking it? > > do you have a debug log showing it happen? > > > > > > > > On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod > > > wrote: > > > > Is there any way to park a channel without causing pre-answer (resulting > is > > a SIP 183 Session Progress)? > > > > > > > > Thanks, > > > > Ron > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From magesh.freeswitch at gmail.com Fri Dec 18 22:18:28 2009 From: magesh.freeswitch at gmail.com (Magesh R) Date: Sat, 19 Dec 2009 11:48:28 +0530 Subject: [Freeswitch-users] Deleting event name in Filter Message-ID: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> Dear All, I have filter a Event like the following in Perl. $IVR->{_esl}->filter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); But I don't how to delete that filter by using filter method. Because 'filter' method accepts only two arguments. Could you any one tell me a way to do it? Thanks, Magesh. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/1d4582b5/attachment-0001.html From scottferri09 at gmail.com Sat Dec 19 10:30:57 2009 From: scottferri09 at gmail.com (Scott Fernandez) Date: Sun, 20 Dec 2009 00:00:57 +0530 Subject: [Freeswitch-users] Third Party device connectivity from Freeswitch Message-ID: Hi, Is there a way to have the Freeswitch to route the calls to physical device/phone rather than just routing the calls to soft phones like Xlite?. If available, What sort of settings that are required in Freeswitch to communicate with third party applications/hardwares (like PBX) so that the calls be switched to physical devices from Freeswitch? Can anyone help me in this regard?. Thanks, Scott. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/012b982d/attachment-0001.html From ron.freeswitch at mcleodnet.com Sat Dec 19 14:34:36 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 14:34:36 -0800 Subject: [Freeswitch-users] Deleting event name in Filter In-Reply-To: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> References: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> Message-ID: Would this work? $IVR->{_esl}->send("filter delete Event-Name CHANNEL_EXECUTE_COMPLETE"); Ron ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Magesh R Sent: Friday, December 18, 2009 10:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Deleting event name in Filter Dear All, ???? I have filter a Event like the following in Perl. ?$IVR->{_esl}->filter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); But I don't how to delete that filter by using filter method. Because 'filter' method accepts? only two arguments. Could you any one tell me a way to do it? Thanks, Magesh. -- This email was Anti Virus checked by Astaro Security Gateway. http://www.astaro.com From frank at carmickle.com Sat Dec 19 14:37:59 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sat, 19 Dec 2009 17:37:59 -0500 Subject: [Freeswitch-users] Third Party device connectivity from Freeswitch In-Reply-To: References: Message-ID: <20091219223759.GI31924@base.carmickle.com> On Sun, Dec 20, Scott Fernandez wrote: > Hi, > > Is there a way to have the Freeswitch to route the calls to physical > device/phone rather than just routing the calls to soft phones like Xlite?. Very much so. It all depends on what you want to do. > If available, What sort of settings that are required in Freeswitch to > communicate with third party applications/hardwares (like PBX) so that the > calls be switched to physical devices from Freeswitch? Have a read through http://wiki.freeswitch.org/wiki/Interop_List HTH --FC From freeswitch at aastral.net Sat Dec 19 15:16:28 2009 From: freeswitch at aastral.net (Bill W.) Date: Sat, 19 Dec 2009 18:16:28 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D0222.7060609@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> Message-ID: <4B2D5ECC.4060209@aastral.net> Hey Metik, Yes. Well, actually, I can have the cidr in two places in the directory. >From what I understand the cidr= parmeter is used in conjunction with the apply-inbound-acl parameter in the sofia profile to just allow someone to make calls from a certain IP without authenticating. And from what I understand the auth-acl= parameter is used to restrict a user to a particular cidr, but the user has to authenticate as well. *The second feature is the one I want to use.* I want to force users to authenticate, but only allow that authentication from a particular cidr as an added measure against toll fraud. And this appears to be causing the issue. Because once I specify the auth-acl parameter in the directory, sofia-reg enforces that acl. And unfortunately it's using the IP of the proxy, not of the user-agent. I looked in sofia.c and found this comment: /* * if network_ip is a proxy allowed to send calls, check for auth * ip header and see if it matches against the inbound acl */ And this coincides with my testing. I have in my profile. I have my proxy sending the X-AUTH-IP header (verified with tcpdump). And yet the REGISTER is still being denied. So it appears that the apply-proxy-acl is set up to work with the apply-inbound-acl ( to allow users from an IP without authenticating) But that hasn't been carried over to sofia_reg.c, which appears to simply check the IP of who FreeSWITCH is talking to against the auth-acl cidr specified in the directory. (Line 1926) So I guess the question is, is my analysis correct? Thoughts anyone? Thanks, Bill Metik wrote: > Bill, > > I think you would add this to the user profile in the directory. The > "brian.xml" example (located in ${confdir}/directory/) provided with the > default/sample configuration files demonstrates how to to do this by > introducing a "cidr" attribute to the the "user" element. > > Example: > > > > > > > > > > > > "http://wiki.freeswitch.org/wiki/Acl" contains some great info > (including a relevant example). > > -metik > From ron.freeswitch at mcleodnet.com Sat Dec 19 17:29:31 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 17:29:31 -0800 Subject: [Freeswitch-users] Difference between ESL execute() and executeAsync() Message-ID: I don't notice any different in behavior between execute() and executeAsync(). I was expecting that executeAsync() would return right-away, and that execute() would only return after the specified application runs to completion (CHANNEL_EXECUTE_COMPLETE event). Running the sample app below, I see the "About to call execute(playback)" and "returned" displayed one right-after the other, even though the file being played takes about 4 minutes to play-out. Do I have this wrong, or is there something incorrect in my app? APP: #!/usr/bin/php events('plain', 'CHANNEL_STATE'); $eventSocket->filter('channel-state', 'CS_ROUTING'); // Wait for new call attempts while($eventSocket->connected()){ $event = $eventSocket->recvEvent(); $serializedBody = $event->serialize(); $listOfLines = toArrayOfLines($serializedBody); $nameValuePairs = toArrayOfNameValuePairs($listOfLines); $uuid = $nameValuePairs['Caller-Unique-ID']; printf("New call from uuid: $uuid\n"); // answer the caller and play announcement $eventSocket->execute('answer', Null ,$uuid); printf("About to call execute(playback)\n"); $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); printf("returned\n"); } ?> DIALPLAN: From ron.freeswitch at mcleodnet.com Sat Dec 19 19:16:03 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 19:16:03 -0800 Subject: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() In-Reply-To: References: Message-ID: Here's the ES network trace: Content-Length: 1502 Content-Type: text/event-plain Event-Name: CHANNEL_STATE Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-19%2019%3A12%3A09 Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT Event-Date-Timestamp: 1261278729767397 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1024 Channel-State: CS_ROUTING Channel-State-Number: 2 Channel-Name: sofia/internal/699%40192.168.100.132 Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: ringing Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 699 Caller-Dialplan: XML Caller-Caller-ID-Name: Ron%20Soft%20Phone Caller-Caller-ID-Number: 699 Caller-Network-Addr: 192.168.100.3 Caller-Destination-Number: 444 Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Caller-Source: mod_sofia Caller-Context: mytest Caller-Channel-Name: sofia/internal/699%40192.168.100.132 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1261278729764077 Caller-Channel-Created-Time: 1261278729764077 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: answer execute-app-arg: Content-Type: command/reply Reply-Text: +OK sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: playback execute-app-arg: /tmp/ann.wav Content-Type: command/reply Reply-Text: +OK > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Saturday, December 19, 2009 5:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Difference between ESL execute() > andexecuteAsync() > > I don't notice any different in behavior between execute() and > executeAsync(). I was expecting that executeAsync() would return > right-away, and that execute() would only return after the specified > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > Running the sample app below, I see the "About to call execute(playback)" > and "returned" displayed one right-after the other, even though the file > being played takes about 4 minutes to play-out. > > Do I have this wrong, or is there something incorrect in my app? > > APP: > #!/usr/bin/php > require_once "ESL.php"; > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > $eventSocket->events('plain', 'CHANNEL_STATE'); > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > // Wait for new call attempts > while($eventSocket->connected()){ > $event = $eventSocket->recvEvent(); > $serializedBody = $event->serialize(); > $listOfLines = toArrayOfLines($serializedBody); > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > $uuid = $nameValuePairs['Caller-Unique-ID']; > printf("New call from uuid: $uuid\n"); > > // answer the caller and play announcement > $eventSocket->execute('answer', Null ,$uuid); > > printf("About to call execute(playback)\n"); > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > printf("returned\n"); > } > ?> > > > DIALPLAN: > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com From gabe at gundy.org Sat Dec 19 20:09:31 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Dec 2009 21:09:31 -0700 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Message-ID: <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> On Thu, Dec 17, 2009 at 6:14 AM, Hristo Benev wrote: > Just for your information there is a version of nmap for windows. So you can do the test from your desktop. Funny that you assume his desktop is running Windows (maybe it is). I would have guessed that the average person on this list doesn't run Windows on the desktop. But, what do I know? Gabe From jason at jasonjgw.net Sat Dec 19 20:18:31 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 20 Dec 2009 15:18:31 +1100 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> Message-ID: <20091220041831.GA9788@jdc.jasonjgw.net> Gabriel Gunderson wrote: > Funny that you assume his desktop is running Windows (maybe it is). I > would have guessed that the average person on this list doesn't run > Windows on the desktop. But, what do I know? Some of us on the list have never run Windows on anything. It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone via a USB head set, and also handling my Snom 320 SIP phone. From john_re at fastmail.us Sun Dec 20 03:13:17 2009 From: john_re at fastmail.us (john_re) Date: Sun, 20 Dec 2009 03:13:17 -0800 Subject: [Freeswitch-users] Dec 20 Global Freeswitch & All Free SW HW Culture meeting - BerkeleyTIP Message-ID: <1261307597.13829.1350992015@webmail.messagingengine.com> Hi FreeSwitchers, & Anthony Anthony - Thanks for letting me post the monthly announcement here. :) FSers: We are working toward moving to FS from Asterisk. We welcome you to join the BTIP Global VOIP bimonthly meetings, & if you like, help us get the FS sw running on our server. :) ===== A great December Solstice to you & yours. :) JOIN the Global All Free SW, HW, Culture meeting via VOIP Dec 20 Sunday, 12N-3PM (Pacific = UTC-8) = 3P-6P Eastern = 8P-11P UTC [Jan 2009 meetings: 2nd, 17th - mark your calendar] http://sites.google.com/site/berkeleytip/schedule == WATCH some VIDEOS: Mark Shuttleworth Interview - 10.04 Lucid Larynx Learning from Code History , Andreas Zeller Why does my program fail? Your version history might have the answer. Audio Hardware Enablement Session, UbuntuDevelopersSummit in Dallas Distributed Development, UDS in Dallas Splunk, Jeremy Thurgood CLUG Upstart, Stefano Rivera CLUG Interfacing with the real world, Mark Ter Morshuizen, Marc Welz CLUG Accelerating Graphics; Camp KDE 2009 http://sites.google.com/site/berkeleytip/talk-videos == Join the MAILING LIST & tell us which videos you will watch & why: http://groups.google.com/group/BerkTIPGlobal == JOIN the meeting via IRC & VOIP: Come discuss any & everything, & work on your individual or group projects. HOT TOPICS: Ub or KUb 9.10?, Ubuntu 10.04 plans, Android, Python3000 in 2010? Start on the #berkeleytip irc.freenode.net channel, & we'll help you get your VOIP system up & working. For VOIP SW, & connection info, see: http://sites.google.com/site/berkeleytip/remote-attendance Berkeley meeting LOCATION: Watch the website & mail list for latest details, perhaps at the Berkeley Public Library, or a cafe, due to Free Speech Cafe closed for winter break. http://sites.google.com/site/berkeleytip/ == OPPORTUNITIES to VOLUNTEER or learn new JOB SKILLS for 2010: Help set up our: Mailing list, FreeSwitch VOIP server, website http://sites.google.com/site/berkeleytip/opportunities Inquire & discuss at the meeting. == For Forwarding - You are invited to forward this announcement wherever it would be appreciated. From mcampbellsmith at gmail.com Sun Dec 20 03:58:02 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 20 Dec 2009 22:58:02 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: Message-ID: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2001 at 192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks From yehavi.bourvine at gmail.com Sun Dec 20 06:26:07 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 20 Dec 2009 16:26:07 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: 2009/12/17 Yehavi Bourvine > I am trying Audiocodes and Vegastream ATAs, and work with either the > manufacturer or the local representative here. > On SNOM I managed to make it work, and will try Polycom soon (once I manage > to grab one unit from our users...). > > Thanks, __yehavi: > > 2009/12/17 Brian West > >> Also what device are you using? I haven't tested with many so far... >> Polycom, Snom and a few others do TLS (see interop page on wiki) others do >> it wrong. >> >> /b >> >> On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: >> >> You could try ssldump: >> >> http://www.rtfm.com/ssldump/ >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/1295a7eb/attachment.html From JCasale at activenetwerx.com Sun Dec 20 07:58:12 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 20 Dec 2009 15:58:12 +0000 Subject: [Freeswitch-users] fs_cli connection error Message-ID: Trying to setup a new config in the pfSense 1.2.3 final package and when I try to connect to the console I get an auth error? # ./fs_cli -H 10.0.0.1 [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error] I tried to search for docs to indicate where one might set the password for this (it never used to have one) but I could only see docs suggesting to provide one, not set one. There is no .fs_cli_conf anywhere. Socketstat shows it listening on 8021... Thanks! jlc From a.afzali2003 at gmail.com Sun Dec 20 08:08:27 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 20 Dec 2009 19:38:27 +0330 Subject: [Freeswitch-users] Interfacing to RabbitMQ Message-ID: Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. Regards, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/2b55e061/attachment.html From msc at freeswitch.org Sun Dec 20 11:41:01 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 20 Dec 2009 11:41:01 -0800 Subject: [Freeswitch-users] fs_cli connection error In-Reply-To: References: Message-ID: The password is set in conf/autoload_configs/event_socket.conf.xml -MC Sent from my iPhone On Dec 20, 2009, at 7:58 AM, "Joseph L. Casale" wrote: > Trying to setup a new config in the pfSense 1.2.3 final package and > when > I try to connect to the console I get an auth error? > > # ./fs_cli -H 10.0.0.1 > [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting > [Authentication Error] > > I tried to search for docs to indicate where one might set the > password for > this (it never used to have one) but I could only see docs > suggesting to provide > one, not set one. > > There is no .fs_cli_conf anywhere. Socketstat shows it listening on > 8021... > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From andrew at hijacked.us Sun Dec 20 11:46:52 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 20 Dec 2009 14:46:52 -0500 Subject: [Freeswitch-users] Interfacing to RabbitMQ In-Reply-To: References: Message-ID: <20091220194651.GB1956@hijacked.us> On Sun, Dec 20, 2009 at 07:38:27PM +0330, afshin afzali wrote: > Hi, > > I'll appreciate if someone who has a practice in interfacing FreeSWITCH to > RabbitMQ or suggestions could share it to me. > You could try to use mod_erlang_event and the erlang rabbitmq client (in native message passing mode). I've never worked with rabbitMQ however, I just know a little about it. Andrew From a.alalousi at gmail.com Sun Dec 20 12:19:27 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Sun, 20 Dec 2009 20:19:27 +0000 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: References: Message-ID: People, Please do excuse me if this is a FAQ. I've so far not worked out a way to implement IP authentication effectively. I have a number of gateways/end points/...etc. hitting the switch without registration and originating calls to a number of upstreams I have configured. So far, everything is smooth and FREESwitch is legendary in every way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon machine running CentOS and CPU usage is barely a blip. My problem is authentication. I need a method to authenticate end points based only on their IP addresses. I am currently filtering access through the firewalls, but I would really like to delegate this task to FS in prep for an SBC setup I'm working on. If I remove the firewall filters, then anyone is able to get in. I could't workout precisely how ACLs work in FS from the WiKi and the documentation, and I haven't been able to make sense of how Digest functions either. Can anyone shed some light on those two areas ? I would like to really get to the bottom of this and update the WiKi pages with the working setup once done. Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/688baec0/attachment.html From mrene_lists at avgs.ca Sun Dec 20 12:26:08 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 20 Dec 2009 15:26:08 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: References: Message-ID: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> Check out: http://wiki.freeswitch.org/wiki/ACL#Users It'll automatically add users with a cidr= attribute to the ACL list. This way you can set channel variables in the users and use them through your dialplan, all authenticated by ip address. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 20-Dec-09, at 3:19 PM, Ahmed Naji wrote: > People, > > Please do excuse me if this is a FAQ. > > I've so far not worked out a way to implement IP authentication > effectively. I have a number of gateways/end points/...etc. hitting > the switch without registration and originating calls to a number of > upstreams I have configured. > > So far, everything is smooth and FREESwitch is legendary in every > way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon > machine running CentOS and CPU usage is barely a blip. > > My problem is authentication. I need a method to authenticate end > points based only on their IP addresses. I am currently filtering > access through the firewalls, but I would really like to delegate > this task to FS in prep for an SBC setup I'm working on. If I remove > the firewall filters, then anyone is able to get in. > > I could't workout precisely how ACLs work in FS from the WiKi and > the documentation, and I haven't been able to make sense of how > Digest functions either. > > Can anyone shed some light on those two areas ? I would like to > really get to the bottom of this and update the WiKi pages with the > working setup once done. > > Regards, > > Ahmed. > > > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/24e95788/attachment.html From freeswitch-users-list at metik.com Sun Dec 20 13:58:52 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 20 Dec 2009 16:58:52 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D5ECC.4060209@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> <4B2D5ECC.4060209@aastral.net> Message-ID: <4B2E9E1C.8090909@metik.com> Then it would appear that my original suggestion to use mod_xml_curl would be best for now and you may need to offer a bounty for this feature as others have suggested. Based on the sofia related snippets presented--I would assume it would be trivial to implement since most of the functionality is already there it just needs to be enhanced for your purpose. It would also be extremely easy to do this in OpenSIPS as well (using blacklists or avpops). Just so that I understand your dilemna, you want to reject an incoming REGISTER associated with a specific user unless it comes from a fixed location and if it does, you want to simply challenge it as usual to prevent toll fraud? I have found that its best to mitigate an attack at ingress before it even makes it to critical infrastructure (media gateways, application/media servers, etc.). -metik Bill W. wrote: > Hey Metik, > > Yes. Well, actually, I can have the cidr in two places in the directory. > > > > > > >From what I understand the cidr= parmeter is used in conjunction with > the apply-inbound-acl parameter in the sofia profile to just allow > someone to make calls from a certain IP without authenticating. > > And from what I understand the auth-acl= parameter is used to restrict a > user to a particular cidr, but the user has to authenticate as well. > > *The second feature is the one I want to use.* I want to force users to > authenticate, but only allow that authentication from a particular cidr > as an added measure against toll fraud. > > And this appears to be causing the issue. Because once I specify the > auth-acl parameter in the directory, sofia-reg enforces that acl. And > unfortunately it's using the IP of the proxy, not of the user-agent. > > I looked in sofia.c and found this comment: > /* > * if network_ip is a proxy allowed to send calls, check for auth > * ip header and see if it matches against the inbound acl > */ > > And this coincides with my testing. > I have in my > profile. I have my proxy sending the X-AUTH-IP header (verified with > tcpdump). And yet the REGISTER is still being denied. > > So it appears that the apply-proxy-acl is set up to work with the > apply-inbound-acl ( to allow users from an IP without authenticating) > > But that hasn't been carried over to sofia_reg.c, which appears to > simply check the IP of who FreeSWITCH is talking to against the auth-acl > cidr specified in the directory. (Line 1926) > > So I guess the question is, is my analysis correct? > > Thoughts anyone? > > Thanks, > Bill > > > > > > > Metik wrote: > >> Bill, >> >> I think you would add this to the user profile in the directory. The >> "brian.xml" example (located in ${confdir}/directory/) provided with the >> default/sample configuration files demonstrates how to to do this by >> introducing a "cidr" attribute to the the "user" element. >> >> Example: >> >> >> >> >> >> >> >> >> >> >> >> "http://wiki.freeswitch.org/wiki/Acl" contains some great info >> (including a relevant example). >> >> -metik >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sun Dec 20 14:28:06 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:28:06 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Message-ID: You'll need to fix your device to know its IP and it should stop doing that. /b On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote: > Hi! > > I'm sure this is a NAT issue, but I'm not sure what options to use. > > I have a Linksys SPA3102, NAT'd on the internet (remotely) and > connected to my FS on the otherside of the world, which is also > natted. A PAP2T is connected on the same subnet as the FS. The 3102 > registers successfully and a call can be set up from the PAP2 to the > 3102. > > However, after FS receives the Remote SDP the audio stops (ring tone > stops in my case) > > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel > sofia/internal/sip:2001 at 192.168.1.3:56885 entering state > [completing][200] > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: > v=0 > o=- 18490612 18490612 IN IP4 192.168.1.3 > s=- > c=IN IP4 192.168.1.3 > t=0 0 > m=audio 16432 RTP/AVP 2 100 101 > a=rtpmap:2 G726-32/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > I notice that the ip address in the o and c fields indicate a local IP > address. Should this IP address be an external IP address of the 3102 > instead? > > Thanks From freeswitch at skillsaw.com Sun Dec 20 12:53:08 2009 From: freeswitch at skillsaw.com (Gad Bentolila) Date: Sun, 20 Dec 2009 15:53:08 -0500 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Message-ID: <4B2E8EB4.2040105@skillsaw.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/3a32b142/attachment.html From brian at freeswitch.org Sun Dec 20 14:36:14 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:36:14 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <4B2E8EB4.2040105@skillsaw.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> Message-ID: <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: > DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. Welcome to the community. > I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). > > Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. > 1) Setup stun on your remote endpoint (spa3102 in your case) > 2) Add to the directory xml file that describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. > Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? > > I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. > > Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/f773fada/attachment.html From brian at freeswitch.org Sun Dec 20 14:45:23 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:45:23 -0600 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: > I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: > > Peer did not provide X.509 Certificate > I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? > > BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. > > Thanks! __Yehavi: From brian at freeswitch.org Sun Dec 20 14:46:47 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:46:47 -0600 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <20091220041831.GA9788@jdc.jasonjgw.net> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> <20091220041831.GA9788@jdc.jasonjgw.net> Message-ID: The funny part is... it won't matter. Their are times when people post questions or issues and its well into debugging the issue before we realize "oh, you're on windows?". For the most part the windows installer is one of the most popular files on our website. /b On Dec 19, 2009, at 10:18 PM, Jason White wrote: > Gabriel Gunderson wrote: > >> Funny that you assume his desktop is running Windows (maybe it is). I >> would have guessed that the average person on this list doesn't run >> Windows on the desktop. But, what do I know? > > Some of us on the list have never run Windows on anything. > > It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone > via a USB head set, and also handling my Snom 320 SIP phone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/21b28f46/attachment.html From mcampbellsmith at gmail.com Sun Dec 20 15:54:43 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 21 Dec 2009 10:54:43 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> Message-ID: <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> Thanks Brian and Gad, I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102. These are the options that I have set (according to the 3102 manual). ? Handle VIA received: yes ? Handle VIA rport: yes ? Insert VIA received: yes ? Insert VIA rport: yes ? Substitute VIA Addr: yes ? Send Resp To Src Port: yes ? STUN Enable: Choose yes. ? STUN Server: stun.freeswitch.org I assume that is all is needed? On Mon, Dec 21, 2009 at 9:36 AM, Brian West wrote: > > On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: > > DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a > grain of salt. > > Welcome to the community. > > I have a similar setup (and problem) - the wiki documentation refers to it > as "double nat". Like you, my FS and client are behind different NATs and I > can register my remote endpoint and make calls (in my case, to the the FS > demo ivr at 5000). > > Since your external endpoint (spa3102) is registering, you've likely setup > your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). > > Your endpoint need only insert rport and FreeSWITCH will do the right thing. > > > 1) Setup stun on your remote endpoint (spa3102 in your case) > 2) Add value="NDLB-connectile-dysfunction"/> to the directory xml file that > describes your spa3102 endpoint > > The device supports STUN also its highly recommended your device know how to > overcome its own NAT. ?I personally do not believe its the registrars place > to overcome an endpoints?nat... puts undue burden on the registar. > > Option 1 worked for me right away (eyebeam in my case) and, as expected, the > remote sdp had the correct (remote) IP address, since the endpoint is using > stun to correctly identify its IP address to FS. However, option 2 has not > made a difference (for me). Is it just me or is it strange that SIP works > without stun, but RTP doesn't? > > I guess I've been spoiled by the way Asterisk handles NAT and was hopeful > that?NDLB-connectile-dysfunction would behave similarly, so I wouldn't have > to tell users to setup stun on their clients.?Maybe a FS user with some > experience with this type of NAT setup and these settings can help. I'd be > interested in knowing how to correctly setup remote NATted endpoints without > stun - or, at least, hear from someone that this setting works for them > without stun. > > Anyway, hope this helps you with your SPA3102. > > Bottom line is enable rport and use stun on the SPA and it'll just work. > /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Sun Dec 20 18:39:54 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 18:39:54 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan Message-ID: <26868725.post@talk.nabble.com> Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26868725.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Sun Dec 20 19:47:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Sun, 20 Dec 2009 19:47:25 -0800 Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <87f2f3b90912201947i46483334kd69938de0446c456@mail.gmail.com> On Sun, Dec 20, 2009 at 6:39 PM, Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be > put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > This functionality already exists in the default dialplan and sample directory entries, assuming that you are using authorization. First off, look in 1000.xml (or any of the other sample user files) for this variable declaration: For any user whom you wish to restrict to local or domestic calling only just remove the 'international' from the list: Now when that user registers and makes calls he/she won't have 'international' in the ${toll_allow} channel variable. Something like this in your dialplan could handle both cases: Now that I've typed all that, I should go back and ask: are you using digest authorization? Or are you using an ACL to let your callers in? Anyway, hopefully the above example will give you some ideas. -MC > Thanks, > Edmar > -- > View this message in context: > http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26868725.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/c9c70b29/attachment.html From darklion11 at yahoo.com Sun Dec 20 21:17:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 21:17:55 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan Message-ID: <26869283.post@talk.nabble.com> Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26869283p26869283.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Dec 20 21:55:36 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 21:55:36 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <26870199.post@talk.nabble.com> Not actually for now... I commented first the ACL restrictions... Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > > Thanks, > Edmar > -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26870199.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Sun Dec 20 22:08:34 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Sun, 20 Dec 2009 22:08:34 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: <1261375714756-4197038.post@n2.nabble.com> Hi, I got it working. Can somebody explain me this error: 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: [Missing local host]. Also I am confused about heartbeat rate. Is enable_heartbeat_events=5 setting the heartbeat to 5? Thank you in advance, Amarakeerthi S wrote: > > Dear Sir, > > I have successfully installed freeSWITCH and it works fine in passthrough > mode. I installed nibblebill and it deduct money from the accounts > database > and it works fine. but I have two problems. > > 1. Calls can be initiated even though there is a minus value in accounts > database > > 2. Calls doesn't hangup when it goes to minus values. > > Any answers are greatly appreciated. > > This is my dialplan: > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Sun Dec 20 22:14:50 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 21 Dec 2009 16:14:50 +1000 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: <1261375714756-4197038.post@n2.nabble.com> References: <1261375714756-4197038.post@n2.nabble.com> Message-ID: what did you have to change, to get this working ? Jay On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S wrote: > > Hi, > > I got it working. > > Can somebody explain me this error: > > 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: > [Missing local host]. Also I am confused about heartbeat rate. Is > enable_heartbeat_events=5 setting the heartbeat to 5? > > > Thank you in advance, > > > > > Amarakeerthi S wrote: > > > > Dear Sir, > > > > I have successfully installed freeSWITCH and it works fine in passthrough > > mode. I installed nibblebill and it deduct money from the accounts > > database > > and it works fine. but I have two problems. > > > > 1. Calls can be initiated even though there is a minus value in accounts > > database > > > > 2. Calls doesn't hangup when it goes to minus values. > > > > Any answers are greatly appreciated. > > > > This is my dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/0d3b9dda/attachment-0001.html From darklion11 at yahoo.com Sun Dec 20 22:31:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 22:31:13 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <26870380.post@talk.nabble.com> Where should I write this line on the default.xml or in the default category? Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > > Thanks, > Edmar > -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26870380.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From talk2ram at gmail.com Sun Dec 20 23:03:46 2009 From: talk2ram at gmail.com (ram) Date: Sun, 20 Dec 2009 23:03:46 -0800 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091219031649.GA1956@hijacked.us> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> Message-ID: Hi its good to hear any compare document between Vicidial and this project Ram On Fri, Dec 18, 2009 at 7:16 PM, Andrew Thompson wrote: > I've been asked to provide some screenshots, so here's some of the > agent/supervisor interface: > > http://eagle.bsd.st/~andrew/cpxshots/ > > Hopefully the image names are self-explanatory. In the ringing picture, > that URL pop is a configurable URL that can be used to integrate with a > CRM, in my case our own CRM - spicecsm. The URL supports interpolation > for variables like callerid, clientid, call type, etc. > > The supervisor view is a little hard to describe via static images, but > you're able to drag and drop agents into another profile (empty profiles > are hidden when not dragging an agent), drag agents onto an agent to > send them the call, and there's also various right click menus > available. > > Oh, and I forgot to mention this before; this system is in 'live > testing' and the goal is to do a final deployment sometime in January. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/ecc6716d/attachment.html From ron.freeswitch at mcleodnet.com Mon Dec 21 00:48:35 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Mon, 21 Dec 2009 00:48:35 -0800 Subject: [Freeswitch-users] How can I detect an execute failure using ESL? Message-ID: <0A22A70A58F642D0A3B778A11C17A67C@fromage> When I try and perform an operation on a channel which has gone, an error is returned. How can I detect this using the ESL? execute() and sendRecv() always return 0 (zero) regardless of whether the command returns +OK or -ERR. sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533 call-command: execute execute-app-name: hangup execute-app-arg: UNALLOCATED_NUMBER Content-Type: command/reply Reply-Text: -ERR invalid session id [5d09753c-ede7-11de-85c6-27ab474dd533] Thanks, Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/8a916666/attachment.html From Prometheus001 at gmx.net Mon Dec 21 04:02:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 21 Dec 2009 13:02:51 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: References: <4B28D6FD.6010702@gmx.net> <4B2B52D7.9030505@gmx.net> Message-ID: <4B2F63EB.9030608@gmx.net> I just crosschecked the dialplan which is used. We do not anwer the call, we bridge it directly to a PSTN destination. However the Ringing event is not passed to PSTN(A): > PSTN(A)====INVITE===>FS > PSTN(A)<===TRYING===>FS > FS===INVITE==>PSTN(B) > FS<==TRYING===PSTN(B) > FS<==RINGING==PSTN(B) > PSTN(A)<==PROGRESS===FS > FS<===OK======PSTN(B) > FS====ACK====>PSTN(B) > PSTN(A)<===OK========FS > PSTN(A)====ACK======>FS But then I stumbled over the following SOFIA LOOPBACK entry in the logs: 2009-12-21 12:47:00.404145 [DEBUG] switch_core_state_machine.c:351 (sofia/external/06322xxxxxxxxxx at 10.11.12.15) State XCHANGE_MEDIA 2009-12-21 12:47:00.404145 [DEBUG] mod_sofia.c:469 SOFIA LOOPBACK 2009-12-21 12:47:00.404145 [DEBUG] sofia.c:3669 Channel sofia/external/0171xxxxxxx at 10.11.12.15:5060 skipping state [early][183] So I modified the dialplan to temporarily use another Patton GW for outgoing calls, et voil?, I receive a ringing tone at PSTN(A). So I think this is because Freeswitch thinks this is a loopback, because incoming and outgoing gateway is the same. But I due to other restrictions we need the call to pass through the same Patton Gateway to PSTN(B) as we received it from PSTN(A). Is there a chance to tell Freeswitch to not consider this call as a loopback scenario? Best regards Peter Brian West schrieb: > That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. > > You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. > > /b > > On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: > > >> Should I open a JIRA for this? >> >> Best regards >> Peter >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Mon Dec 21 07:03:58 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 09:03:58 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> Message-ID: <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> Can you get me siptraces please. /b On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote: > Thanks Brian and Gad, > > I have stun set and if I do a 'sofia status profile internal', I see > the external IP address of the 3102 ATA, so I assume that stun is > working correctly on the SPA3102. > > These are the options that I have set (according to the 3102 manual). > > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server: stun.freeswitch.org > > I assume that is all is needed? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/833b0c89/attachment.html From senakahks at gmail.com Mon Dec 21 07:36:31 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Mon, 21 Dec 2009 07:36:31 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: <1261375714756-4197038.post@n2.nabble.com> Message-ID: I have changed the changed the dialplan little bit (thank to the people at IRC #freeswitch) as follows. Here I don't understand enable_heartbeat_events=5. It may be the heartbeat rate. Also I am getting this error in FS cli 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: [Missing local host] On Tue, Dec 22, 2009 at 12:12 AM, jay binks [via freeswitch-users] wrote: > what did you have to change, to get this working ? > Jay > > On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S <[hidden email]> wrote: >> >> Hi, >> >> I got it working. >> >> Can somebody explain me this error: >> >> 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS >> ERROR: >> [Missing local host]. Also I am confused about heartbeat rate. Is >> enable_heartbeat_events=5 ?setting the heartbeat to 5? >> >> >> Thank you in advance, >> >> >> >> >> Amarakeerthi S wrote: >> > >> > Dear Sir, >> > >> > I have successfully installed freeSWITCH and it works fine in >> > passthrough >> > mode. I installed nibblebill and it deduct money from the accounts >> > database >> > and it works fine. but I have two problems. >> > >> > 1. Calls can be initiated even though there is a minus value in accounts >> > database >> > >> > 2. Calls doesn't hangup when it goes to minus values. >> > >> > Any answers are greatly appreciated. >> > >> > This is my dialplan: >> > >> > >> > >> > >> > ? >> > ? ? >> > ? ? >> > ? >> > >> > ? >> > >> > >> > >> > >> > >> > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> >> > >> > >> > >> > >> > >> > This is the configuration file; >> > >> > >> > ? >> > ? ? >> > >> > ? ? >> > >> > >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > >> > ? >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > [hidden email] >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > View message @ > http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4198003.html > To unsubscribe from Re: Mod nibblebill deduct money but no hangup at zero > and can call without money in database, click here. > -- View this message in context: http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4198998.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/b6828d92/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 21 07:42:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 09:42:30 -0600 Subject: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() In-Reply-To: References: Message-ID: <191c3a030912210742y108f764et83581b50bd2bd60b@mail.gmail.com> if you run the socket in async mode, every call to execute is async if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod wrote: > Here's the ES network trace: > > Content-Length: 1502 > Content-Type: text/event-plain > Event-Name: CHANNEL_STATE > Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-19%2019%3A12%3A09 > Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT > Event-Date-Timestamp: 1261278729767397 > Event-Calling-File: switch_channel.c > Event-Calling-Function: switch_channel_perform_set_running_state > Event-Calling-Line-Number: 1024 > Channel-State: CS_ROUTING > Channel-State-Number: 2 > Channel-Name: sofia/internal/699%40192.168.100.132 > Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f > Call-Direction: inbound > Presence-Call-Direction: inbound > Answer-State: ringing > Channel-Read-Codec-Name: PCMU > Channel-Read-Codec-Rate: 8000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Caller-Username: 699 > Caller-Dialplan: XML > Caller-Caller-ID-Name: Ron%20Soft%20Phone > Caller-Caller-ID-Number: 699 > Caller-Network-Addr: 192.168.100.3 > Caller-Destination-Number: 444 > Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f > Caller-Source: mod_sofia > Caller-Context: mytest > Caller-Channel-Name: sofia/internal/699%40192.168.100.132 > Caller-Profile-Index: 1 > Caller-Profile-Created-Time: 1261278729764077 > Caller-Channel-Created-Time: 1261278729764077 > Caller-Channel-Answered-Time: 0 > Caller-Channel-Progress-Time: 0 > Caller-Channel-Progress-Media-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > > > sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f > call-command: execute > execute-app-name: answer > execute-app-arg: > > > Content-Type: command/reply > Reply-Text: +OK > > > sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f > call-command: execute > execute-app-name: playback > execute-app-arg: /tmp/ann.wav > > > Content-Type: command/reply > Reply-Text: +OK > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > > Sent: Saturday, December 19, 2009 5:30 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Difference between ESL execute() > > andexecuteAsync() > > > > I don't notice any different in behavior between execute() and > > executeAsync(). I was expecting that executeAsync() would return > > right-away, and that execute() would only return after the specified > > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > > > Running the sample app below, I see the "About to call execute(playback)" > > and "returned" displayed one right-after the other, even though the file > > being played takes about 4 minutes to play-out. > > > > Do I have this wrong, or is there something incorrect in my app? > > > > APP: > > #!/usr/bin/php > > > require_once "ESL.php"; > > > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > > $eventSocket->events('plain', 'CHANNEL_STATE'); > > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > > > // Wait for new call attempts > > while($eventSocket->connected()){ > > $event = $eventSocket->recvEvent(); > > $serializedBody = $event->serialize(); > > $listOfLines = toArrayOfLines($serializedBody); > > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > > > $uuid = $nameValuePairs['Caller-Unique-ID']; > > printf("New call from uuid: $uuid\n"); > > > > // answer the caller and play announcement > > $eventSocket->execute('answer', Null ,$uuid); > > > > printf("About to call execute(playback)\n"); > > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > > printf("returned\n"); > > } > > ?> > > > > > > DIALPLAN: > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > > This email was Anti Virus checked by Astaro Security Gateway. > > http://www.astaro.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/c369e10a/attachment.html From andrew at hijacked.us Mon Dec 21 08:07:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 21 Dec 2009 11:07:08 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> Message-ID: <20091221160708.GC1956@hijacked.us> On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > Hi > > its good to hear > > any compare document between Vicidial and this project > No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew From jbr at consiglia.dk Mon Dec 21 08:13:21 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Mon, 21 Dec 2009 17:13:21 +0100 Subject: [Freeswitch-users] mod_xml_curl and gateways Message-ID: I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/ed0eb650/attachment.html From mrene_lists at avgs.ca Mon Dec 21 08:21:04 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 21 Dec 2009 11:21:04 -0500 Subject: [Freeswitch-users] mod_xml_curl and gateways In-Reply-To: References: Message-ID: Hi, All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files. This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 21-Dec-09, at 11:13 AM, Jon Bruel wrote: > I wonder if it is possible to define common gateways (not user > specific gateways) by xml_curl, and if so, the bindings and syntax > to use? > > All the best /Jon > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/bec4ea70/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 21 08:22:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 10:22:54 -0600 Subject: [Freeswitch-users] How can I detect an execute failure using ESL? In-Reply-To: <0A22A70A58F642D0A3B778A11C17A67C@fromage> References: <0A22A70A58F642D0A3B778A11C17A67C@fromage> Message-ID: <191c3a030912210822x14699f7fid2e9e077c31102c8@mail.gmail.com> the latest version returns an event with that data in it similar to the api method. On Mon, Dec 21, 2009 at 2:48 AM, Ron McLeod wrote: > When I try and perform an operation on a channel which has gone, an error > is returned. How can I detect this using the ESL? execute() and sendRecv() > always return 0 (zero) regardless of whether the command returns *+OK* or > *?ERR*. > > > > sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533 > > call-command: execute > > execute-app-name: hangup > > execute-app-arg: UNALLOCATED_NUMBER > > > > Content-Type: command/reply > > *Reply-Text: -ERR invalid session id > [5d09753c-ede7-11de-85c6-27ab474dd533]* > > > > Thanks, > > Ron > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/7b48361d/attachment.html From anthony.minessale at gmail.com Mon Dec 21 08:26:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 10:26:12 -0600 Subject: [Freeswitch-users] mod_xml_curl and gateways In-Reply-To: References: Message-ID: <191c3a030912210826k7d2ba4e5n400883b366d1fc4d@mail.gmail.com> same exact syntax only put the in the sofia profile On Mon, Dec 21, 2009 at 10:13 AM, Jon Bruel wrote: > I wonder if it is possible to define common gateways (not user specific > gateways) by xml_curl, and if so, the bindings and syntax to use? > > > > All the best /Jon > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/13312702/attachment.html From freeswitch at aastral.net Mon Dec 21 08:47:17 2009 From: freeswitch at aastral.net (Bill W) Date: Mon, 21 Dec 2009 11:47:17 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2E9E1C.8090909@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> <4B2D5ECC.4060209@aastral.net> <4B2E9E1C.8090909@metik.com> Message-ID: <4B2FA695.3040803@aastral.net> Hey Metik, Thank you so much for your assistance on this issue. I really appreciate it. Yes I agree with you on the mod_xml_curl solution. However, as I was starting to pursue that, I ran into another issue. It appears as though I don't have access to any variables in the xml_curl POST that contain the IP of the UA. The only two variables with IPs (other than the switch IP) are: sip_contact_host=192.168.0.100 and ip=64.135.119.105 where the .105 is my proxy. :( Do you know of any way to get additional variables into the xml_curl POST? As far as my current use case, yes, you understand my needs correctly, with one slight modification, I want to use the IP acl+Auth with both REGISTERs and INVITEs. And yes, I agree with you that it is better to mitigate at the border, but I don't have that kind of infrastructure available yet. So do you have any other suggestions on a workaround with the xml_curl issue? Or should I include that with my bounty? Thanks, Bill Metik wrote: > Then it would appear that my original suggestion to use mod_xml_curl > would be best for now and you may need to offer a bounty for this > feature as others have suggested. Based on the sofia related snippets > presented--I would assume it would be trivial to implement since most of > the functionality is already there it just needs to be enhanced for your > purpose. It would also be extremely easy to do this in OpenSIPS as well > (using blacklists or avpops). > > Just so that I understand your dilemna, you want to reject an incoming > REGISTER associated with a specific user unless it comes from a fixed > location and if it does, you want to simply challenge it as usual to > prevent toll fraud? > > I have found that its best to mitigate an attack at ingress before it > even makes it to critical infrastructure (media gateways, > application/media servers, etc.). > > -metik > > Bill W. wrote: >> Hey Metik, >> >> Yes. Well, actually, I can have the cidr in two places in the directory. >> >> >> >> >> >> >From what I understand the cidr= parmeter is used in conjunction with >> the apply-inbound-acl parameter in the sofia profile to just allow >> someone to make calls from a certain IP without authenticating. >> >> And from what I understand the auth-acl= parameter is used to restrict a >> user to a particular cidr, but the user has to authenticate as well. >> >> *The second feature is the one I want to use.* I want to force users to >> authenticate, but only allow that authentication from a particular cidr >> as an added measure against toll fraud. >> >> And this appears to be causing the issue. Because once I specify the >> auth-acl parameter in the directory, sofia-reg enforces that acl. And >> unfortunately it's using the IP of the proxy, not of the user-agent. >> >> I looked in sofia.c and found this comment: >> /* >> * if network_ip is a proxy allowed to send calls, check for auth >> * ip header and see if it matches against the inbound acl >> */ >> >> And this coincides with my testing. >> I have in my >> profile. I have my proxy sending the X-AUTH-IP header (verified with >> tcpdump). And yet the REGISTER is still being denied. >> >> So it appears that the apply-proxy-acl is set up to work with the >> apply-inbound-acl ( to allow users from an IP without authenticating) >> >> But that hasn't been carried over to sofia_reg.c, which appears to >> simply check the IP of who FreeSWITCH is talking to against the auth-acl >> cidr specified in the directory. (Line 1926) >> >> So I guess the question is, is my analysis correct? >> >> Thoughts anyone? >> >> Thanks, >> Bill >> >> >> >> >> >> >> Metik wrote: >> >>> Bill, >>> >>> I think you would add this to the user profile in the directory. The >>> "brian.xml" example (located in ${confdir}/directory/) provided with the >>> default/sample configuration files demonstrates how to to do this by >>> introducing a "cidr" attribute to the the "user" element. >>> >>> Example: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> "http://wiki.freeswitch.org/wiki/Acl" contains some great info >>> (including a relevant example). >>> >>> -metik >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Mon Dec 21 09:02:33 2009 From: freeswitch at aastral.net (Bill W) Date: Mon, 21 Dec 2009 12:02:33 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> Message-ID: <4B2FAA29.4010405@aastral.net> I recently added an overview to this wiki page to help make things more clear as to which ACL you need for different purposes. http://wiki.freeswitch.org/wiki/ACL#Overview Thanks, Bill W. Mathieu Rene wrote: > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > It'll automatically add users with a cidr= attribute to the ACL list. > This way you can set channel variables in the users and use them through > your dialplan, all authenticated by ip address. > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca From msc at freeswitch.org Mon Dec 21 09:48:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Dec 2009 09:48:44 -0800 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091221160708.GC1956@hijacked.us> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> <20091221160708.GC1956@hijacked.us> Message-ID: <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson wrote: > On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > > Hi > > > > its good to hear > > > > any compare document between Vicidial and this project > > > > No document, but briefly: > > * More focused on inbound than on outbound (at least for the moment) > vicidial is more geared for outbound. > * Handles email in queue (and soon chat), vicidial is only voice. > * wrapup time is per-call not static per-'campaign' > * license is a little more liberal > * can operate as a distributed system > * doesn't need asterisk ;) > Now *that* is a feature worth paying for! ;) Also, I thought you had a community edition vs. a professional edition? If so could you explain the difference? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/6afc69f1/attachment.html From qinglan_zeng at hotmail.com Mon Dec 21 10:16:14 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Mon, 21 Dec 2009 18:16:14 +0000 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: Hello, I noticed some guys had develop the Skype module while there is a policy from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means)" , which means once you use Skype unlimited calls plan into PBX, Skype will frozen your account without any money return. That's a big risk for anybody to use Skype unlimited call plan. My question is how do we avoid such kind of risk? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 42, Issue 193 To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 08:21:08 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 09:42:30 -0600 Subject: Re: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() if you run the socket in async mode, every call to execute is async if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod wrote: Here's the ES network trace: Content-Length: 1502 Content-Type: text/event-plain Event-Name: CHANNEL_STATE Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-19%2019%3A12%3A09 Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT Event-Date-Timestamp: 1261278729767397 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1024 Channel-State: CS_ROUTING Channel-State-Number: 2 Channel-Name: sofia/internal/699%40192.168.100.132 Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: ringing Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 699 Caller-Dialplan: XML Caller-Caller-ID-Name: Ron%20Soft%20Phone Caller-Caller-ID-Number: 699 Caller-Network-Addr: 192.168.100.3 Caller-Destination-Number: 444 Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Caller-Source: mod_sofia Caller-Context: mytest Caller-Channel-Name: sofia/internal/699%40192.168.100.132 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1261278729764077 Caller-Channel-Created-Time: 1261278729764077 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: answer execute-app-arg: Content-Type: command/reply Reply-Text: +OK sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: playback execute-app-arg: /tmp/ann.wav Content-Type: command/reply Reply-Text: +OK > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Saturday, December 19, 2009 5:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Difference between ESL execute() > andexecuteAsync() > > I don't notice any different in behavior between execute() and > executeAsync(). I was expecting that executeAsync() would return > right-away, and that execute() would only return after the specified > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > Running the sample app below, I see the "About to call execute(playback)" > and "returned" displayed one right-after the other, even though the file > being played takes about 4 minutes to play-out. > > Do I have this wrong, or is there something incorrect in my app? > > APP: > #!/usr/bin/php > require_once "ESL.php"; > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > $eventSocket->events('plain', 'CHANNEL_STATE'); > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > // Wait for new call attempts > while($eventSocket->connected()){ > $event = $eventSocket->recvEvent(); > $serializedBody = $event->serialize(); > $listOfLines = toArrayOfLines($serializedBody); > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > $uuid = $nameValuePairs['Caller-Unique-ID']; > printf("New call from uuid: $uuid\n"); > > // answer the caller and play announcement > $eventSocket->execute('answer', Null ,$uuid); > > printf("About to call execute(playback)\n"); > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > printf("returned\n"); > } > ?> > > > DIALPLAN: > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 --??????-- From: andrew at hijacked.us To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 11:07:08 -0500 Subject: Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > Hi > > its good to hear > > any compare document between Vicidial and this project > No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew --??????-- From: jbr at consiglia.dk To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 17:13:21 +0100 Subject: [Freeswitch-users] mod_xml_curl and gateways I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon --??????-- From: mrene_lists at avgs.ca To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 11:21:04 -0500 Subject: Re: [Freeswitch-users] mod_xml_curl and gateways Hi, All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files. This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 21-Dec-09, at 11:13 AM, Jon Bruel wrote: I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ ?Windows Live ???????Messenger2009???? http://www.windowslive.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/10f83f1c/attachment-0001.html From itamar at ispbrasil.com.br Mon Dec 21 10:37:38 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Mon, 21 Dec 2009 16:37:38 -0200 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: 2009/12/21 ??? : > Hello, > > I noticed some guys had develop the Skype module while there is a policy > from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used > by one person only and is not to be shared with any other user (whether via > a PBX, call centre, computer or any other means)" , which means once you use > Skype unlimited calls plan into PBX, Skype will frozen your account without > any money return. That's a big risk for anybody to use Skype unlimited call > plan. > > My question is how do we avoid such kind of risk? > > Thanks > Daniel Zeng the best answer is don't use skype. ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From brian at freeswitch.org Mon Dec 21 10:43:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 12:43:25 -0600 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: So says the man with his Skype username in his sig! :P /b On Dec 21, 2009, at 12:37 PM, Itamar Reis Peixoto wrote: > the best answer is don't use skype. > > > > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/041fac2e/attachment.html From JCasale at activenetwerx.com Mon Dec 21 11:56:59 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 19:56:59 +0000 Subject: [Freeswitch-users] sound rpms Message-ID: So the spec from trunk says "Soundfiles are moving into a separate spec" but I can't find this spec anywhere in svn? Anyone know where it is? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/bfe3a72a/attachment.html From mike at jerris.com Mon Dec 21 12:25:59 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Dec 2009 15:25:59 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: Message-ID: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Working on it, moving the repos around to do this right... http://jira.freeswitch.org/browse/FSBUILD-218 Mike On Dec 21, 2009, at 2:56 PM, Joseph L. Casale wrote: > So the spec from trunk says ?Soundfiles are moving into a separate spec? > but I can?t find this spec anywhere in svn? > > Anyone know where it is? > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/0a2de892/attachment.html From JCasale at activenetwerx.com Mon Dec 21 12:49:28 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 20:49:28 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >Working on it, moving the repos around to do this right... > >http://jira.freeswitch.org/browse/FSBUILD-218 > >Mike Thanks, Is this known to not work with non root builds? It errored out after creating some messy hierarchies with the actual variable calls, instead of their values? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/91a52343/attachment.html From mike at jerris.com Mon Dec 21 13:00:35 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Dec 2009 16:00:35 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: This is a total work in progress that has not even merged into tree. So it is not "known" to work or not work anywhere. Patches to correct issues are welcome. Mike On Dec 21, 2009, at 3:49 PM, Joseph L. Casale wrote: > >Working on it, moving the repos around to do this right... > > > >http://jira.freeswitch.org/browse/FSBUILD-218 > > > >Mike > > Thanks, Is this known to not work with non root builds? It errored out after creating some > messy hierarchies with the actual variable calls, instead of their values? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/ba0d897f/attachment.html From JCasale at activenetwerx.com Mon Dec 21 13:04:51 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 21:04:51 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >Thanks, Is this known to not work with non root builds? It errored out after creating some >messy hierarchies with the actual variable calls, instead of their values? Actually, I tried on a lab vm as root in the typical dirs. and got the same result: ... `./us/callie/time/48000/hours.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/hours.wav' `./us/callie/time/48000/oclock.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/oclock.wav' `./us/callie/time/48000/mon-6.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/mon-6.wav' error: Bad exit status from /var/tmp/rpm-tmp.19453 (%install) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.19453 (%install) From stevendt at primrosebank.net Mon Dec 21 13:25:36 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 21 Dec 2009 21:25:36 -0000 Subject: [Freeswitch-users] Building on Windows References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com><20091217171527.GA16380@hijacked.us> Message-ID: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Hi Mike, OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! Most of the "warnings" in the build have been cleared, I notice that the "error" in the build of mod_opal has gone with mod_opal not being in the preconfigured build list now and also that the number of warnings on VS2008 has reduced since I've been playing with this over the past few days and SVN versions. There does not seem to be anything major wrong, but as you requested, I have raised a Jira (FSBUILD-221) that you might care to take a look at please ? I attached a copy of the output from the build run and highlighted in bold in the RTF file the warnings that are generated. There are a host of warnings due to the use of /analyze which is appears not to be supported by the Express compiler. Most significant (though still trivial) are the warnings of some type conversion problems and some "indirection" errors. As I said, these don't seem to be too much of a problem, but you may care to take a look when you have time. One more thing, when I do subsequent builds, are there any pre-build steps that I need to take ? When I tried rebuilding previously, there seemed to be some directories that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) that were flagged as problems - do I just ignore these warnings or unprotect and/or delete the directories before rebuilding ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 17, 2009 6:14 PM Subject: Re: [Freeswitch-users] Building on Windows On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/fccae920/attachment-0001.html From andrew at hijacked.us Mon Dec 21 14:36:59 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 21 Dec 2009 17:36:59 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> <20091221160708.GC1956@hijacked.us> <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> Message-ID: <20091221223659.GE1956@hijacked.us> On Mon, Dec 21, 2009 at 09:48:44AM -0800, Michael Collins wrote: > On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson wrote: > > > On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > > > Hi > > > > > > its good to hear > > > > > > any compare document between Vicidial and this project > > > > > > > No document, but briefly: > > > > * More focused on inbound than on outbound (at least for the moment) > > vicidial is more geared for outbound. > > * Handles email in queue (and soon chat), vicidial is only voice. > > * wrapup time is per-call not static per-'campaign' > > * license is a little more liberal > > * can operate as a distributed system > > * doesn't need asterisk ;) > > > Now *that* is a feature worth paying for! ;) > > Also, I thought you had a community edition vs. a professional edition? If > so could you explain the difference? I've managed to avoid that thus far, I suspect that something like an outbound campaign manager (which could be implemented as just another media type) might be something to fall under that sort of split, but right now the release includes everything we've got (including an integration module that's probably of limited use to anyone else - but its a good example of how to build your own). Andrew From larclap at yahoo.com Mon Dec 21 14:50:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 21 Dec 2009 14:50:11 -0800 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <4B2FAA29.4010405@aastral.net> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> Message-ID: <007701ca828f$f414f9b0$dc3eed10$@com> Bill, Thanks for your ACL Overview. Perhaps you can help me understand more clearly. If you include the "local-network-acl" and "apply-inbound-acl" params in the sip_profiles and setup the list for "localnet.auto" in acl.conf.xml, does this mean you do not have to include the cidr attribute for individual extensions in the directory/default folder? Is "apply-inbound-acl" supposed to exist in both internal and external profiles while "apply-inbound-acl" is only in the internal? Thanks, Lars > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users- > bounces at lists.freeswitch.org] On Behalf Of Bill W > Sent: Monday, December 21, 2009 9:03 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Authenticating end points by IP > > I recently added an overview to this wiki page to help make things more > clear as to which ACL you need for different purposes. > > http://wiki.freeswitch.org/wiki/ACL#Overview > > Thanks, > Bill W. > > > Mathieu Rene wrote: > > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > > > It'll automatically add users with a cidr= attribute to the ACL list. > > This way you can set channel variables in the users and use them through > > your dialplan, all authenticated by ip address. > > > > Cheers, > > > > Mathieu Rene > > Avant-Garde Solutions Inc > > Office: + 1 (514) 664-1044 x100 > > Cell: +1 (514) 664-1044 x200 > > mrene at avgs.ca > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Mon Dec 21 15:53:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Dec 2009 15:53:49 -0800 Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26870380.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> <26870380.post@talk.nabble.com> Message-ID: <87f2f3b90912211553r75204938y2ea845020db1ac01@mail.gmail.com> On Sun, Dec 20, 2009 at 10:31 PM, Edmar Cruz wrote: > > Where should I write this line > > > > data="misc/you-are-not-authorized.wav"/> > > > > > > > on the default.xml or in the default category? > You can put it in default.xml or in an xml file in conf/dialplan/default/ -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/08d5c351/attachment.html From JCasale at activenetwerx.com Mon Dec 21 16:03:48 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 22 Dec 2009 00:03:48 +0000 Subject: [Freeswitch-users] Variables for install directories Message-ID: Searching through the wiki for any indication as to what if any variables exist for the install location in that I can leverage in a script. Can anyone point me along, I can't seem to find anything. I want to place a shell script in /opt/freeswitch/scripts that needs a reference to a conf file that a binary it runs is calling. So now I have in two places hardcoded paths that I was hoping to avoid, in the dialplan and in the shell script. When either of these is run, does there exist something like and the same for use inside the shell script? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/e5c757ef/attachment.html From jerry.richards at teotech.com Mon Dec 21 16:24:06 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 21 Dec 2009 16:24:06 -0800 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> Message-ID: <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the WARNING in the Freeswitch log. However, it still behaves the same way. That is, the internal callee rings for about 12 seconds, then stops ringing, and the PSTN caller just hears ringback for about 60 seconds and is not given the opportunity to leave voice mail. In contrast, an internal-to-internal call will go to voice mail after 30 seconds. I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe driver (or Freeswitch) setting that would correct this? Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, November 03, 2009 2:23 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WARNING On Inbound Call Question can you try the same thing with the latest trunk or pre-release tarball. On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards wrote: I have my Freeswitch server with an installed Sangoma A101D card. Most everything works okay, however, when I get an inbound call from the PSTN, I see the following warning show up in the log. Additionally, the caller (on the PSTN) does not hear ringback, and if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. Here are the two warnings: [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA Here is the log of the warning upon an inbound call: freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on 1:1 from DOWN to RING 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig [START] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound channel OpenZAP/1:1/5384 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT going to sleep 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 CHANNEL ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 OpenZAP/1:1/5384 Standard ROUTING 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing 4253813176->5384 in context default Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] Dialplan: OpenZAP/1:1/5384 Action set(open=true) Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] destination_number(5384) =~ /^9(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] destination_number(5384) =~ /^(5380)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] destination_number(5384) =~ /^\*8$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~ /^870$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] destination_number(5384) =~ /^9001$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] destination_number(5384) =~ /^9000$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^779$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] destination_number(5384) =~ /^80(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] destination_number(5384) =~ /^81(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] destination_number(5384) =~ /^82(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] destination_number(5384) =~ /^83(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: OpenZAP/1:1/5384 Action answer() Dialplan: OpenZAP/1:1/5384 Action sleep(1000) Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING going to sleep 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 (OpenZAP/1:1/5384) State EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 CHANNEL EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 OpenZAP/1:1/5384 Standard EXECUTE EXECUTE OpenZAP/1:1/5384 set(open=true) 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [open]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a 53132) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/4253813176/5384) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 3132) EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [transfer_ringback]=[local_stream://moh] EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [call_timeout]=[30] EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [hangup_after_bridge]=[true] EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [continue_on_fail]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-call_return/5384/4253813176) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 a53132) EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [called_party_callgroup]=[techsupport] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 d89a53132) EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=5384 at 192.168.72.141] 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:5384 at 192.168.72.163:5060 [9e7b8fae-6194-430c-951b-948ebd2c2a3b] 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 OpenZAP/1:1/5384 receive message [PROGRESS] 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 from RING to PROGRESS 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [PROGRESS] 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer OpenZAP/1:1/5384! 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [proceeding][180] 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:5384 at 192.168.72.163:5060! 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 receive message [TRANSCODING_NECESSARY] freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/4052d79c/attachment-0001.html From jerry.richards at teotech.com Mon Dec 21 16:44:49 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 21 Dec 2009 16:44:49 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time Message-ID: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry From brian at freeswitch.org Mon Dec 21 16:52:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 18:52:20 -0600 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> Message-ID: <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> You know that warning is meaningless. Search the archives we have talked about this to no end it seems. And I'm sure Moy fixed this. /b On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: > Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the WARNING in the Freeswitch log. However, it still behaves the same way. That is, the internal callee rings for about 12 seconds, then stops ringing, and the PSTN caller just hears ringback for about 60 seconds and is not given the opportunity to leave voice mail. In contrast, an internal-to-internal call will go to voice mail after 30 seconds. > > I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe driver (or Freeswitch) setting that would correct this? > > Best Regards, > Jerry > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/05e3af79/attachment.html From jeff at jefflenk.com Mon Dec 21 20:18:40 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 22:18:40 -0600 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com><20091217171527.GA16380@hijacked.us>, , <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff From: stevendt at primrosebank.net To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 21:25:36 +0000 Subject: Re: [Freeswitch-users] Building on Windows Hi Mike, OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! Most of the "warnings" in the build have been cleared, I notice that the "error" in the build of mod_opal has gone with mod_opal not being in the preconfigured build list now and also that the number of warnings on VS2008 has reduced since I've been playing with this over the past few days and SVN versions. There does not seem to be anything major wrong, but as you requested, I have raised a Jira (FSBUILD-221) that you might care to take a look at please ? I attached a copy of the output from the build run and highlighted in bold in the RTF file the warnings that are generated. There are a host of warnings due to the use of /analyze which is appears not to be supported by the Express compiler. Most significant (though still trivial) are the warnings of some type conversion problems and some "indirection" errors. As I said, these don't seem to be too much of a problem, but you may care to take a look when you have time. One more thing, when I do subsequent builds, are there any pre-build steps that I need to take ? When I tried rebuilding previously, there seemed to be some directories that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) that were flagged as problems - do I just ignore these warnings or unprotect and/or delete the directories before rebuilding ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 17, 2009 6:14 PM Subject: Re: [Freeswitch-users] Building on Windows On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Hotmail: Free, trusted and rich email service. http://clk.atdmt.com/GBL/go/171222984/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/54ad2aca/attachment.html From jeff at jefflenk.com Mon Dec 21 20:22:09 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 20:22:09 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: <1261455729581-4201954.post@n2.nabble.com> Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff Dave Stevenson wrote: > > Hi Mike, > > OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! > > Most of the "warnings" in the build have been cleared, I notice that the > "error" in the build of mod_opal has gone with mod_opal not being in the > preconfigured build list now and also that the number of warnings on > VS2008 has reduced since I've been playing with this over the past few > days and SVN versions. > > There does not seem to be anything major wrong, but as you requested, I > have raised a Jira (FSBUILD-221) that you might care to take a look at > please ? > > I attached a copy of the output from the build run and highlighted in bold > in the RTF file the warnings that are generated. > There are a host of warnings due to the use of /analyze which is appears > not to be supported by the Express compiler. > > Most significant (though still trivial) are the warnings of some type > conversion problems and some "indirection" errors. > > As I said, these don't seem to be too much of a problem, but you may care > to take a look when you have time. > > One more thing, when I do subsequent builds, are there any pre-build steps > that I need to take ? > When I tried rebuilding previously, there seemed to be some directories > that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) > that were flagged as problems - do I just ignore these warnings or > unprotect and/or delete the directories before rebuilding ? > > regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, December 17, 2009 6:14 PM > Subject: Re: [Freeswitch-users] Building on Windows > > > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > > > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > > Hi, > > > > I'm probably going to regret this - I'm not sure that I'll be able > to do this without a lot of pain (nothing to do with FS - more my lack of > ability with Visual Studio), but......, I want to try building FreeSwitch > from source rather than using the pre-built binaries. I have a couple of > initial questions that, hopefully, someone can answer please ? > > > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 > on the horizon for me. > > Having downloaded the SVN, I see there is a VS 2005 Solution, but it > is marked as "Unsupported", although the Wiki says that you only need > VC++2005. > > What does "unsupported" mean in this context ? I guess that support > for VS2005 is being dropped, but is the VS2005 Solution still being > maintained, and if so, for how long? I'd hate to get into the build thing > and then find that I was stalled when VS2005 support was dropped > altogether ? > > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules > stand > a good chance of not having support. > > > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > > > > > 2. The whole SVN thing is new to me but I've worked out that I need > an SVN Client on Windows to work with the source. Can anyone recommend the > best (free) SVN Client for Windows to use with FreeSwitch. I have > installed TortoiseSVN - a Windows Explorer Shell that looks pretty and > seemed to work on my first build but it's not command line based so some > of the tips given in the Wiki like "make current" and "make sounds" may be > more awkward to achieve. Is anyone else using Tortoise and/or can give > some tips on which SVN client to use ? > > > > Tortoise SVN is fine and is probably the de-facto client for windows. > > > > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] > > 3. I built 15979 last night (with VS2005) and got some warnings, > with data type conversion - is this a known issue under Windows ? > > > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > > > > > 4. There was one fatal error in the build of mod_opal (missing file) > > (Some examples of the warnings and the error are shown below :-) > > > > Try with VS 2008 and see if they go away. > > > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > > > 5. How do I specify which options (e.g., mod_flite, to be included > iin the build. > > > > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. > > > > You can adjust this in the configuration managaer > > > 6. How do I build the sounds etc. ? > > > > > The sounds are a subproject too IIRC. > > > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > > > Mike > > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4201954.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Mon Dec 21 20:23:31 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 20:23:31 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: <1261455811463-4201959.post@n2.nabble.com> Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff Dave Stevenson wrote: > > Hi Mike, > > OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! > > Most of the "warnings" in the build have been cleared, I notice that the > "error" in the build of mod_opal has gone with mod_opal not being in the > preconfigured build list now and also that the number of warnings on > VS2008 has reduced since I've been playing with this over the past few > days and SVN versions. > > There does not seem to be anything major wrong, but as you requested, I > have raised a Jira (FSBUILD-221) that you might care to take a look at > please ? > > I attached a copy of the output from the build run and highlighted in bold > in the RTF file the warnings that are generated. > There are a host of warnings due to the use of /analyze which is appears > not to be supported by the Express compiler. > > Most significant (though still trivial) are the warnings of some type > conversion problems and some "indirection" errors. > > As I said, these don't seem to be too much of a problem, but you may care > to take a look when you have time. > > One more thing, when I do subsequent builds, are there any pre-build steps > that I need to take ? > When I tried rebuilding previously, there seemed to be some directories > that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) > that were flagged as problems - do I just ignore these warnings or > unprotect and/or delete the directories before rebuilding ? > > regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, December 17, 2009 6:14 PM > Subject: Re: [Freeswitch-users] Building on Windows > > > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > > > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > > Hi, > > > > I'm probably going to regret this - I'm not sure that I'll be able > to do this without a lot of pain (nothing to do with FS - more my lack of > ability with Visual Studio), but......, I want to try building FreeSwitch > from source rather than using the pre-built binaries. I have a couple of > initial questions that, hopefully, someone can answer please ? > > > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 > on the horizon for me. > > Having downloaded the SVN, I see there is a VS 2005 Solution, but it > is marked as "Unsupported", although the Wiki says that you only need > VC++2005. > > What does "unsupported" mean in this context ? I guess that support > for VS2005 is being dropped, but is the VS2005 Solution still being > maintained, and if so, for how long? I'd hate to get into the build thing > and then find that I was stalled when VS2005 support was dropped > altogether ? > > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules > stand > a good chance of not having support. > > > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > > > > > 2. The whole SVN thing is new to me but I've worked out that I need > an SVN Client on Windows to work with the source. Can anyone recommend the > best (free) SVN Client for Windows to use with FreeSwitch. I have > installed TortoiseSVN - a Windows Explorer Shell that looks pretty and > seemed to work on my first build but it's not command line based so some > of the tips given in the Wiki like "make current" and "make sounds" may be > more awkward to achieve. Is anyone else using Tortoise and/or can give > some tips on which SVN client to use ? > > > > Tortoise SVN is fine and is probably the de-facto client for windows. > > > > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] > > 3. I built 15979 last night (with VS2005) and got some warnings, > with data type conversion - is this a known issue under Windows ? > > > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > > > > > 4. There was one fatal error in the build of mod_opal (missing file) > > (Some examples of the warnings and the error are shown below :-) > > > > Try with VS 2008 and see if they go away. > > > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > > > 5. How do I specify which options (e.g., mod_flite, to be included > iin the build. > > > > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. > > > > You can adjust this in the configuration managaer > > > 6. How do I build the sounds etc. ? > > > > > The sounds are a subproject too IIRC. > > > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > > > Mike > > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4201959.html Sent from the freeswitch-users mailing list archive at Nabble.com. From john at acsol.net Mon Dec 21 16:15:22 2009 From: john at acsol.net (john at acsol.net) Date: Mon, 21 Dec 2009 17:15:22 -0700 Subject: [Freeswitch-users] Multitenant dialplans Message-ID: <4b300f9a.313.2c10.1142196461@acsol.net> I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to "sofia/internal/sip:extersion at public-IP:translated-port"; however when I log into "Company1" with the phones, it tries "sofia/internal/dialed-extension at Company1" ... I also get "User not Registered". The dialplans are the same either way. Any ideas? Thanks John From Prometheus001 at gmx.net Tue Dec 22 03:20:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Dec 2009 12:20:39 +0100 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time In-Reply-To: References: Message-ID: <4B30AB87.3060909@gmx.net> Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a 200 > OK with only the PCMU codec. This looks incorrect. The audio call > previously negotiated to the speex/16000 codec, and the re-INVITE from the > caller added the H263-1998 codec. If I re-attempt to start video at the > caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the complete > scenario: establishing audio call, first failed start video attempt, and > second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Tue Dec 22 03:40:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Dec 2009 12:40:11 +0100 Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf Message-ID: <4B30B01B.30809@gmx.net> Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter From scott.torr.fs at letterboxes.org Tue Dec 22 06:57:05 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Wed, 23 Dec 2009 01:57:05 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? Message-ID: <1261493825.21085.1351311647@webmail.messagingengine.com> ubuntu-8.04.3-server-amd64.iso (update/upgrade) FreeSWITCH Version 1.0.trunk (15787) skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb mod_skypiax (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) fs>console loglevel 7 If I dial 501 from from a sip phone using "inband" dtmf I can see the dtmf tones being detected and decoded by fs in the debug log. If however I use a pstn phone and dial my skypeIN telephone number the call comes into fs via skypiax but when I generate dtmf tones on the phone they are not detected or decoded by fs. If I take the record_session file and spectrum analyze the recorded tones appear to be within spec. Can anybody suggest why this is not working for me? Is the correct sample rate being used in libteletone_detect.c? Does the Goertzel algorithm work for other sample rates other than 8000hz? I'm not sure why I can not get this to work? regards, Scott Torr From anthony.minessale at gmail.com Tue Dec 22 07:12:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Dec 2009 09:12:30 -0600 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time In-Reply-To: References: Message-ID: <191c3a030912220712o5d028687hb3be922eb56a47f6@mail.gmail.com> Can you repeat that same trace with latest trunk? On Mon, Dec 21, 2009 at 6:44 PM, Jerry Richards wrote: > > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a 200 > OK with only the PCMU codec. This looks incorrect. The audio call > previously negotiated to the speex/16000 codec, and the re-INVITE from the > caller added the H263-1998 codec. If I re-attempt to start video at the > caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the complete > scenario: establishing audio call, first failed start video attempt, and > second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/99bcf8eb/attachment.html From brian at freeswitch.org Tue Dec 22 07:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 09:13:54 -0600 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4b300f9a.313.2c10.1142196461@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> Message-ID: <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: > I have Freeswitch setup and working as a single tenant > system mostly using the default configuration. Trying to > convert to a multitenant environment, I have used both the > Multi-tenant and Multiple Companies wiki's. I get the phone > to register, can call out using the external profile to a > ITSP, can call music on hold; however I can not call other > users in the company. > It appears that when logged in with single company and > default context it sucessfully calls other internal phones > with bridge to > "sofia/internal/sip:extersion at public-IP:translated-port"; > however when I log into "Company1" with the phones, it tries > "sofia/internal/dialed-extension at Company1" ... I also get > "User not Registered". The dialplans are the same either > way. > > Any ideas? > > Thanks > John > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Dec 22 07:21:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Dec 2009 09:21:16 -0600 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261493825.21085.1351311647@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> Message-ID: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> add "start_dtmf" app to your dialplan before bridge to start the inband dtmf detector. On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr wrote: > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > FreeSWITCH Version 1.0.trunk (15787) > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > mod_skypiax > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > fs>console loglevel 7 > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > dtmf tones being detected and decoded by fs in the debug log. > > > If however I use a pstn phone and dial my skypeIN telephone number the > call comes into fs via skypiax but when I generate dtmf tones on the > phone they are not detected or decoded by fs. > > If I take the record_session file and spectrum analyze the recorded > tones appear to be within spec. > > > Can anybody suggest why this is not working for me? > > > Is the correct sample rate being used in libteletone_detect.c? > Does the Goertzel algorithm work for other sample rates other than > 8000hz? > > > I'm not sure why I can not get this to work? > > > > regards, > Scott Torr > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/e5a9da29/attachment-0001.html From gmaruzz at celliax.org Tue Dec 22 07:25:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 22 Dec 2009 16:25:21 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261493825.21085.1351311647@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> Message-ID: <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> It is probably because mod_skypiax does not analize incoming audio looking for dtmf, because the "normal" call from a Skype client peer sends *both* inband and out of band (signaling) dtmf. So, I choose to only detect out of band (signaling) dtmfs, and ignore possible inband dtmfs (in the audio flow), so to have the most reliable source (signaling) and spare cpu (not analizing the incoming audio). Never tought you can receive calls from skypeIN with inband dtmfs... Open a Jira for this, I'll think about. Also, let me know your toughts... -giovanni On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr wrote: > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > FreeSWITCH Version 1.0.trunk (15787) > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > mod_skypiax > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > ? > ? ? > ? ? ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > ? ? > ? > > > > fs>console loglevel 7 > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > dtmf tones being detected and decoded by fs in the debug log. > > > If however I use a pstn phone and dial my skypeIN telephone number the > call comes into fs via skypiax but when I generate dtmf tones on the > phone they are not detected or decoded by fs. > > If I take the record_session file and spectrum analyze the recorded > tones appear to be within spec. > > > Can anybody suggest why this is not working for me? > > > Is the correct sample rate being used in libteletone_detect.c? > Does the Goertzel algorithm work for other sample rates other than > 8000hz? > > > I'm not sure why I can not get this to work? > > > > regards, > Scott Torr > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Tue Dec 22 07:26:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 22 Dec 2009 16:26:01 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> Message-ID: <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> do as anthm say :-) On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale wrote: > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf > detector. > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > wrote: >> >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> FreeSWITCH Version 1.0.trunk (15787) >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> mod_skypiax >> >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> >> >> ? >> ? ? >> ? ?> >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> ? ? >> ? >> >> >> >> fs>console loglevel 7 >> >> >> If I dial 501 from from a sip phone using "inband" dtmf I can see the >> dtmf tones being detected and decoded by fs in the debug log. >> >> >> If however I use a pstn phone and dial my skypeIN telephone number the >> call comes into fs via skypiax but when I generate dtmf tones on the >> phone they are not detected or decoded by fs. >> >> If I take the record_session file and spectrum analyze the recorded >> tones appear to be within spec. >> >> >> Can anybody suggest why this is not working for me? >> >> >> Is the correct sample rate being used in libteletone_detect.c? >> Does the Goertzel algorithm work for other sample rates other than >> 8000hz? >> >> >> I'm not sure why I can not get this to work? >> >> >> >> regards, >> Scott Torr >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jerry.richards at teotech.com Tue Dec 22 08:02:21 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 22 Dec 2009 08:02:21 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail Message-ID: <19FBB8C038E64C1DB92B2842AF8BCF01@greyhawk.tonecommander.com> I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry From john at acsol.net Tue Dec 22 08:16:13 2009 From: john at acsol.net (John) Date: Tue, 22 Dec 2009 09:16:13 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> Message-ID: <4B30F0CD.8040703@acsol.net> Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1004 at company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: > The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. > > /b > > On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: > > >> I have Freeswitch setup and working as a single tenant >> system mostly using the default configuration. Trying to >> convert to a multitenant environment, I have used both the >> Multi-tenant and Multiple Companies wiki's. I get the phone >> to register, can call out using the external profile to a >> ITSP, can call music on hold; however I can not call other >> users in the company. >> It appears that when logged in with single company and >> default context it sucessfully calls other internal phones >> with bridge to >> "sofia/internal/sip:extersion at public-IP:translated-port"; >> however when I log into "Company1" with the phones, it tries >> "sofia/internal/dialed-extension at Company1" ... I also get >> "User not Registered". The dialplans are the same either >> way. >> >> Any ideas? >> >> Thanks >> John >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jerry.richards at teotech.com Tue Dec 22 08:33:00 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 22 Dec 2009 08:33:00 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time In-Reply-To: <4B30AB87.3060909@gmx.net> References: <4B30AB87.3060909@gmx.net> Message-ID: No. The following lines is commented out (internal.xml): Thanks, Jerry -----Original Message----- From: Peter P GMX [mailto:Prometheus001 at gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a > 200 OK with only the PCMU codec. This looks incorrect. The audio > call previously negotiated to the speex/16000 codec, and the re-INVITE > from the caller added the H263-1998 codec. If I re-attempt to start > video at the caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the > complete > scenario: establishing audio call, first failed start video attempt, > and second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From brian at freeswitch.org Tue Dec 22 10:04:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 12:04:30 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] "a1-has" param in gateway setting In-Reply-To: <179369.24879.qm@web110203.mail.gq1.yahoo.com> References: <179369.24879.qm@web110203.mail.gq1.yahoo.com> Message-ID: I'm not too sure you can put an a1-hash on outbound auth. /b On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote: > Hi, > > Does any body know or has test the "a1-hash" parameter with gateway > setting? I am not sure if it is even allowed. I have the following > gateway setting but when the freeswitch starts up it simply ignores this > provider without any error message or attempt to register in the log > file. Thank you for your help in advance. > > > > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/1f3a4443/attachment.html From john at acsol.net Tue Dec 22 10:21:32 2009 From: john at acsol.net (John) Date: Tue, 22 Dec 2009 11:21:32 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B30F0CD.8040703@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> Message-ID: <4B310E2C.8020501@acsol.net> One point of clarification, currently all the phones are behind NAT, so it appears that when the phones are in a Non-multitenant scenario, they use SIP:dialed_number at IP-address-of-their-gateway. On 12/22/2009 9:16 AM, John wrote: > Thanks Brian. I did have both force-register-domain and > force-register-db-domain commented in both the internal.xml and > internal-ipv6.xml. The phones appear to register to the company1 domain, > as shown in sofia status profile company1; however I have noticed that > when I try to make a call to another a phone in the same domain, the > system is trying to call sofia/internal/1004 at company1 -- this is when we > get the message, user not registered. If I can the phones to just > register to the IP address of the machine, they call fine and is shows > sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan > problem? In both cases I am just using the sample dialplan. > > > > > On 12/22/2009 8:13 AM, Brian West wrote: > >> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. >> >> /b >> >> On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: >> >> >> >>> I have Freeswitch setup and working as a single tenant >>> system mostly using the default configuration. Trying to >>> convert to a multitenant environment, I have used both the >>> Multi-tenant and Multiple Companies wiki's. I get the phone >>> to register, can call out using the external profile to a >>> ITSP, can call music on hold; however I can not call other >>> users in the company. >>> It appears that when logged in with single company and >>> default context it sucessfully calls other internal phones >>> with bridge to >>> "sofia/internal/sip:extersion at public-IP:translated-port"; >>> however when I log into "Company1" with the phones, it tries >>> "sofia/internal/dialed-extension at Company1" ... I also get >>> "User not Registered". The dialplans are the same either >>> way. >>> >>> Any ideas? >>> >>> Thanks >>> John >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From lcm at marshap.com Tue Dec 22 10:50:04 2009 From: lcm at marshap.com (Larry Marshall) Date: Tue, 22 Dec 2009 10:50:04 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests Message-ID: <005101ca8337$9335ffb0$b9a1ff10$@com> I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml "external_sip_ip=stun:stun.freeswitch.org" On the original box in vars.xml "external_sip_port=5090" but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/be50aadd/attachment-0001.html From msc at freeswitch.org Tue Dec 22 11:14:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 11:14:36 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <005101ca8337$9335ffb0$b9a1ff10$@com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> Message-ID: <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall wrote: > I have set up a second FreeSWITCH box on the same LAN. I have v16018 > installed on it and have changed nothing. > > > > I configured a Polycom phone to register one of its four lines to this > second box, but it does not register. When looking at the console, there is > no activity. However, there is SIP activity on the box which I have captured > via ngrep. It looks like the phone is sending out REGISTER requests but > there is no response. The request on the pastebin repeats forever, with only > the timestamp varying. > > On the new box do "sofia status" - does the internal profile exist? > > > Is the problem that there are two FreeSWITCHes? Any suggestions on how I > can make it work? > > > > On the original and the new box in vars.xml "external_sip_ip=stun: > stun.freeswitch.org" > > On the original box in vars.xml "external_sip_port=5090" but in the new it > is 5080. > > > > Do I need to hardcode the external_sip_ip addresses in both boxes? > > > > http://pastebin.freeswitch.org/11600 > > > > Thanks Lars > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/641d2a22/attachment.html From codecomplete at free.fr Tue Dec 22 11:24:55 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 22 Dec 2009 11:24:55 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all Message-ID: <26892767.post@talk.nabble.com> Hello I'm running "1.0.trunk (15841)" on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues: 1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away ("the person on extension 1001 is not available") 2. When calling GS from XLite, the GS phone doesn't even ring. FWIW, the phones seem to have registered OK: freeswitch at internal> sofia status profile internal Registrations: ======================================================== Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1001 at 192.168.0.7 Contact: "Freeswitch" Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account: 1001 at 192.168.0.7 Call-ID: 3f6d4ebebd5e829f at 192.168.0.9 User: 1003 at 192.168.0.7 Contact: "user" Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account: 1003 at 192.168.0.7 ======================================================== Has someone seen this type of behavior? Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26892767.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 22 11:30:54 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 22 Dec 2009 11:30:54 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: <26893059.post@talk.nabble.com> I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Tue Dec 22 11:35:55 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:35:55 +0200 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: Try tracing the calls from both sides with TCPDUMP or enable siptrace on FreeSwitch. I guess this will give you some clue. __Yehavi: 2009/12/22 Fred-145 > > Hello > > I'm running "1.0.trunk (15841)" on Linux CentOS with a the default > settings. > After succesfully connecting a Windows PC running XLite (EyeBeam, really) > and a GrandStream IP phone to Freeswitch, I try to make calls, but am > having > the following issues: > > 1. When calling XLite from GS, XLite rings, but when I pick up the call, > the > caller is sent to voice-mail right away ("the person on extension 1001 is > not available") > 2. When calling GS from XLite, the GS phone doesn't even ring. > > FWIW, the phones seem to have registered OK: > > freeswitch at internal> sofia status profile internal > Registrations: > ======================================================== > Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. > User: 1001 at 192.168.0.7 > Contact: "Freeswitch" > > Agent: eyeBeam release 1104a stamp 54437 > Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) > Host: centos.workgroup > IP: 192.168.0.1 > Port: 41380 > Auth-User: 1001 > Auth-Realm: 192.168.0.7 > MWI-Account: 1001 at 192.168.0.7 > > Call-ID: 3f6d4ebebd5e829f at 192.168.0.9 > User: 1003 at 192.168.0.7 > Contact: "user" > > Agent: Grandstream BT120 1.1.0.3 > Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) > Host: centos.workgroup > IP: 192.168.0.9 > Port: 5060 > Auth-User: 1003 > Auth-Realm: 192.168.0.7 > MWI-Account: 1003 at 192.168.0.7 > ======================================================== > > Has someone seen this type of behavior? > > Thanks for any hint. > -- > View this message in context: > http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26892767.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/f7721fb2/attachment.html From yehavi.bourvine at gmail.com Tue Dec 22 11:39:15 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:39:15 +0200 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26893059.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <26893059.post@talk.nabble.com> Message-ID: It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 > > I found the cause for #2: The GS phone was still configured to use NAT, > even > though both XLite and GS are located in the same, private LAN. Unchecking > this on the GS phone solved the issue. > > But I'm still having issue #1, regardless of which phone is calling or > being > called: When the phone answers the call, I'm sent automatically to > voice-mail. Could it be codec-related, or something like that? > > Thank you. > -- > View this message in context: > http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/1f9cbe0e/attachment.html From yehavi.bourvine at gmail.com Tue Dec 22 11:43:05 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:43:05 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> Message-ID: My distro is fedora 10 with all the current patches. SSLwatch fails to build and it seems more than a trivial change to make it work; however, it seems that the error message from Freeswitch tells it all... Is there any special debug statement in Freeswitch to see more about its TLS negotations? Thanks, __Yehavi: 2009/12/21 Brian West > You have to watch it with TLS. Make sure your distro didn't mess up your > SSL libs due to the recent vulnerability found. I havn't tested with my > polycom in a few weeks but it was working on my Polycom after I uploaded the > ca cert and marked it as trusted/used on the phone. > > /b > > On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: > > > I am trying now to set a Polycom to work with FreeSwitch and TLS. I have > a Polycom-501 which does not have an internal certificate, thus only one-way > certificate validation is needed. I've downloaded the root certificate to he > Polyciom, and Freeswitch gives me the following error: > > > > Peer did not provide X.509 Certificate > > I understand that it tries to do mutual authentication which is not > possible in this case. How can I tell FreeSwitch to ignore the client's > certificate? > > > > BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and > Yealink. > > > > Thanks! __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/357b25a9/attachment-0001.html From msc at freeswitch.org Tue Dec 22 11:44:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 11:44:14 -0800 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: References: <26892767.post@talk.nabble.com> Message-ID: <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine wrote: > Try tracing the calls from both sides with TCPDUMP or enable siptrace on > FreeSwitch. I guess this will give you some clue. > > __Yehavi: > Additionally, turn on debugging on the console and capture that output. If you use fs_cli it has debug output turned on by default. Pastebin that output and post the link in this thread. If you happen to look at the traces and figure it out then please let us know. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/9eaa9a83/attachment.html From larclap at yahoo.com Tue Dec 22 11:46:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 22 Dec 2009 11:46:11 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> Message-ID: <006c01ca833f$6a43d890$3ecb89b0$@com> Yes, the internal profile exists. Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 192.168.10.25 alias internal ALIASED ============================================================================ ===================== 3 profiles 1 alias From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 22, 2009 11:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall wrote: I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. On the new box do "sofia status" - does the internal profile exist? Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml "external_sip_ip=stun:stun.freeswitch.org" On the original box in vars.xml "external_sip_port=5090" but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/c351afd4/attachment.html From mastermind202 at gmail.com Tue Dec 22 11:51:25 2009 From: mastermind202 at gmail.com (mm_202) Date: Tue, 22 Dec 2009 14:51:25 -0500 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 In-Reply-To: <200912171305.57498.yivzhenko@mksat.net> References: <200912171305.57498.yivzhenko@mksat.net> Message-ID: <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> Yuriy, The FS wiki has examples of how to control the BLF/MWI using events. I had no problem getting to work with my GXP2020. Let me know if you want some direct code examples. -- MM. On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko wrote: > Hallo All! > I need information about setup BLF on GXP2010/2020 phones with Freeswitch. > I search in Freeswitch Wiki and maillist archives but find no usable > information. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From a.alalousi at gmail.com Tue Dec 22 12:24:43 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 22 Dec 2009 20:24:43 +0000 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <4B2FAA29.4010405@aastral.net> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> Message-ID: Excellent work and answers. Thanks gentlemen. I'm firing off a new thread re: codecs et. al. Have a great Christmas and a wonderful, prosperous New Year. Regards, Ahmed. 2009/12/21 Bill W > I recently added an overview to this wiki page to help make things more > clear as to which ACL you need for different purposes. > > http://wiki.freeswitch.org/wiki/ACL#Overview > > Thanks, > Bill W. > > > Mathieu Rene wrote: > > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > > > It'll automatically add users with a cidr= attribute to the ACL list. > > This way you can set channel variables in the users and use them through > > your dialplan, all authenticated by ip address. > > > > Cheers, > > > > Mathieu Rene > > Avant-Garde Solutions Inc > > Office: + 1 (514) 664-1044 x100 > > Cell: +1 (514) 664-1044 x200 > > mrene at avgs.ca > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/53ba0867/attachment.html From a.alalousi at gmail.com Tue Dec 22 12:55:41 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 22 Dec 2009 20:55:41 +0000 Subject: [Freeswitch-users] Codecs and things Message-ID: Hello people, Can someone please clear the following ambiguities with codecs: 1. Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard 2. When an A-leg has negotiated a pass-through media codec, can the B-leg be transcoded into a non-pass-through codec, and vice-versa ? think A-leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say 3. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? 4. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/4f015cf1/attachment.html From msc at freeswitch.org Tue Dec 22 14:17:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 14:17:23 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5pre10 is now available Message-ID: <87f2f3b90912221417l3b4b7f00pe7a4c7775ce2d85a@mail.gmail.com> It's upgrade-and-test time! The new release announcement is on the main FreeSWITCH page: http://www.freeswitch.org/node/224 Please update, test, and report back bugs and questions. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/118432aa/attachment-0001.html From msc at freeswitch.org Tue Dec 22 14:20:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 14:20:32 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <006c01ca833f$6a43d890$3ecb89b0$@com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> Message-ID: <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: > Yes, the internal profile exists. > > > > Name Type > Data State > > > ================================================================================================= > > internal profile sip:mod_sofia at 192.168.10.25:5060 > RUNNING (0) > > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > > external profile sip:mod_sofia at 192.168.10.25:5080 > RUNNING (0) > > example.com gateway sip:joeuser at example.com > NOREG > > 192.168.10.25 alias > internal ALIASED > > > ================================================================================================= > > 3 profiles 1 alias > > > I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/b126118d/attachment.html From vinuth.madinur at gmail.com Tue Dec 22 14:28:03 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 03:58:03 +0530 Subject: [Freeswitch-users] Choosing a Codec. Message-ID: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> Hi, I am playing a file to a landline number. the format of the file is as follows: [root at static-host var]# file message.wav message.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz In my vars.xml file I have used the following codec prefs: However, when freeswitch plays it, it always chooses the L16 at 8000hz codec. I'm not understanding why this is so. EXECUTE sofia/external/5135692990 at 208.78.161.197 playback(/var/message.wav) 2009-12-22 17:16:57.357048 [DEBUG] switch_ivr_play_say.c:1135 Codec Activated L16 at 8000hz 1 channels 20ms 2009-12-22 17:17:30.777182 [DEBUG] switch_ivr_play_say.c:1429 done playing file My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. Have I configured it wrong or does this transcoding always happen? Thanks, Vinuth. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/3d272e5d/attachment.html From kristoff.bonne at skypro.be Tue Dec 22 14:06:47 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Tue, 22 Dec 2009 23:06:47 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch Message-ID: <4B3142F7.1080600@skypro.be> Hi all, This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating-systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb-audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/498362d4/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/498362d4/attachment.bin From freeswitch at aastral.net Tue Dec 22 14:38:10 2009 From: freeswitch at aastral.net (Bill W) Date: Tue, 22 Dec 2009 17:38:10 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <007701ca828f$f414f9b0$dc3eed10$@com> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> <007701ca828f$f414f9b0$dc3eed10$@com> Message-ID: <4B314A52.8080808@aastral.net> Hello Lars, You can apply any acl to any profile. What you should do really depends on what you want to accomplish. But let's take a simple example. Let's say you want to allow any phone on your internal network (192.168.0.0/24) to connect to your internal profile and make calls without having to provide a password. Then you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from. For your external profile, you may not want to have any ACLs at all, as you may not want to limit which IPs can connect to your switch to send you incoming calls. BUT, you need to make sure the dialplan connected to that external profile doesn't allow anyone to dial numbers that are not hosted on your system without proper authentication or controls. And believe me, people WILL try to do that. I've set up my system to email me whenever this happens and I have logged over 100 attempts to dial international numbers just since December 3rd. Hope this helps, Bill Lars Zeb wrote: > Bill, > > Thanks for your ACL Overview. Perhaps you can help me understand more > clearly. > > If you include the "local-network-acl" and "apply-inbound-acl" params in the > sip_profiles and setup the list for "localnet.auto" in acl.conf.xml, does > this mean you do not have to include the cidr attribute for individual > extensions in the directory/default folder? > > Is "apply-inbound-acl" supposed to exist in both internal and external > profiles while "apply-inbound-acl" is only in the internal? > > Thanks, Lars > From brian at freeswitch.org Tue Dec 22 14:39:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 16:39:06 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> Message-ID: <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: > My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. From vinuth.madinur at gmail.com Tue Dec 22 14:54:47 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 04:24:47 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> Message-ID: <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:09 AM, Brian West wrote: > Why? You don't have to avoid it... why bother? > > /b > > On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: > > > My basic intent is to avoid on-the-fly transcoding, while having a high > quality audio playing on PSTN. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/20e9b85f/attachment.html From brian at freeswitch.org Tue Dec 22 15:11:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 17:11:16 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> Message-ID: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. > > Thanks, > Vinuth. From Mailings at kh-dev.de Tue Dec 22 15:22:30 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Wed, 23 Dec 2009 00:22:30 +0100 Subject: [Freeswitch-users] Make error... Message-ID: Hi all, I just downloaded the newest trunk about 5 minutes ago and I got the following make error on Ubuntu 8.04: gcc -E /usr/src/freeswitch/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive mkdir .libs Compiling src/switch_apr.c ... cc1: warnings being treated as errors src/switch_apr.c: In function 'switch_uuid_parse': src/switch_apr.c:899: warning: control reaches end of non-void function make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make: *** [all] Error 2 Regards, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/c47eed65/attachment-0001.html From jason at jasonjgw.net Tue Dec 22 15:38:29 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Dec 2009 10:38:29 +1100 Subject: [Freeswitch-users] Make error... In-Reply-To: References: Message-ID: <20091222233829.GA8702@jdc.jasonjgw.net> Klaus Hochlehnert wrote: > src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. From Mailings at kh-dev.de Tue Dec 22 16:20:01 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Wed, 23 Dec 2009 01:20:01 +0100 Subject: [Freeswitch-users] Make error... In-Reply-To: <20091222233829.GA8702@jdc.jasonjgw.net> References: <20091222233829.GA8702@jdc.jasonjgw.net> Message-ID: I was on 16031. Now I downloaded 16032 and currently the make is running. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, December 23, 2009 12:38 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make error... Klaus Hochlehnert wrote: > src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rupa at rupa.com Tue Dec 22 16:45:37 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 22 Dec 2009 18:45:37 -0600 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji wrote: > Hello people, > > Can someone please clear the following ambiguities with codecs: > > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy > Media mode, or does FS need to be running in bypass-media ? the Wiki is not > clear in this regard Yes, you can use proxy media, bypass media, or even regular mode if you don't transcode (special for g729). Proxy media is really a special hack that should only be used for T38 passthrough. If you are using it for other purposes, think about it some more.... > When an A-leg has negotiated a pass-through media codec, can the B-leg be > transcoded into a non-pass-through codec, and vice-versa ? think A-leg > incoming with a G.729 codec, and target for B-leg needs to be setup with a > GSM-codec, say That would require transcoding - which can't be done if the codec is pass-through. > Where in the developer's set of documentation are codecs discussed ? I would > like to start porting some code of mine for G.729a/b/ab form a ti DSP > platform to FS. FS lacking full G.729 support is proving quite a hindrance, > and there is no clear direction from the dev community as to when the same > will be available. Incidentally, any news on this effort ? where are we with > code, and what's an ETA for a Beta ? I'd say look at the broadvoice or other simple self-contained codecs are done. Currently the only supported g729 solution is to use a digium board with mod_dahdi_codec. I don't have any info on a software based g729 solution. > On the same lines as (3) above, there is a codec dev template in the source > tree. Again, where can I find documentation relating to this ? the template > has hardly any docs at all. > > Best regards and warm wishes for a Merry Christmas and a great New Year to > one and all. > > Ahmed. > > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From rupa at rupa.com Tue Dec 22 16:47:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 22 Dec 2009 18:47:52 -0600 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B3142F7.1080600@skypro.be> References: <4B3142F7.1080600@skypro.be> Message-ID: Interesting. It would have to do more than just dialtone/dtmf though. Need call control, caller id, etc. What do they ship with it as far as drivers go? On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne wrote: > Hi all, > > > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) > > > What is interesting on this device is that is uses standard USB > device-classes that are by default supported by most operating-systems: > usb-sound and usb-hid. > > > When I connect it to my server (mac mini 3G running debian), the system > automatically recognises these two classes > > [168391.922479] usbcore: registered new interface driver hiddev > [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on > usb-0001:10:1b.1-1 > [168391.939548] usbcore: registered new interface driver usbhid > [168391.943984] usbhid: v2.6:USB HID core driver > [168392.154596] usbcore: registered new interface driver snd-usb-audio > > > And -behold- when I connect a handset in one of the port, I even get a > dialtone and I can sent out DTMF-dialtone which are somehow partly > (But I have no idea what program actually generates this dialtone !!!) > > > > Now, the question: > Any idea if / how this can incorperated into freeswitch? Is there a way > to use this device to connect a phone to freeswitch without having to go > throu a SIP-client first. > > > > Cheerio! Kr. Bonne. > > -- > jabber/gtalk: kristoff at krbonne.net > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From dave at 3c.co.uk Tue Dec 22 19:29:28 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 23 Dec 2009 03:29:28 +0000 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> Message-ID: <1261538968.18473.55.camel@local.freepabx.com> On the other hand, a u-law WAV turned into L16 and then back to u-law to be sent down the line shouldn't suffer any alteration at all - if it does, the there's something wrong with the translation. The quality dropping over time is almost certainly down to something else. Vinuth -can you get a recording to compare with the original? --Dave > If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. > > Can you elaborate on your setup a bit more? > > /b > > On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > > > The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Tue Dec 22 19:44:18 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 22 Dec 2009 22:44:18 -0500 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <9dc4a1670912221944y3c302d37gc466f45a5d60df0f@mail.gmail.com> Have you considered GIPS http://www.gipscorp.com/products/overview.php ? -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/99a38e79/attachment.html From JCasale at activenetwerx.com Tue Dec 22 19:58:30 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 23 Dec 2009 03:58:30 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: >> Am I correct in presuming that Freeswitch will answer a fax from a local zap based user >> just like it does from an FXO port connected to a POTS line? What I hope to do here is >> catch any call made from that extension (the zap based fax machine/user) and push its >> call into the fax module. > > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone > and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that > dials. This extension looks ok. Try it out and let us know how it goes. > -MC Michael, It worked well, there was however a humorous moment: I was testing with my own shell script that simply emailed me directly to my postfix gateway, my exchange server and mua understood the uuencoded attachment so once it started working I modified the script to send to our fax service. Well they didn't understand uuencode so the attachment, a single page tiff, got faxed as 23 pages of binary :) I used mutt with a redirection to a specific muttrc which understands mime encoding which should work everywhere... Thanks for the help, you've made an office full of people happy... jlc From mike at jerris.com Tue Dec 22 20:13:21 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:13:21 -0500 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> Message-ID: <4D7D2B1F-7880-4DDE-8F71-1D6F78B6E212@jerris.com> If this is using prid it also requires the latest drivers from sangoma. I am pretty sure these are just in dev snapshots not release drivers yet. Something 3.5.8.6 or later iirc. Mike On Dec 21, 2009, at 7:52 PM, Brian West wrote: > You know that warning is meaningless. Search the archives we have > talked about this to no end it seems. > > And I'm sure Moy fixed this. > > /b > > On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: > >> Okay, I upgraded to 1.0.5pre9 and tried this test again and I do >> not see the WARNING in the Freeswitch log. However, it still >> behaves the same way. That is, the internal callee rings for about >> 12 seconds, then stops ringing, and the PSTN caller just hears >> ringback for about 60 seconds and is not given the opportunity to >> leave voice mail. In contrast, an internal-to-internal call will >> go to voice mail after 30 seconds. >> >> I put a new 11595 log into the pastebin. Is there some Sangoma >> Wanpipe driver (or Freeswitch) setting that would correct this? >> >> Best Regards, >> Jerry >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/2597ed69/attachment-0001.html From mike at jerris.com Tue Dec 22 20:31:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:31:16 -0500 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: References: Message-ID: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> For the path in the dialplan I don't think we have any right now but file a bug on jira and I can try to add them. As for something in the script itself that is a bit more work but if anyone has a patch to inject some vars into scripts like that it would be a nice addition. Mike On Dec 21, 2009, at 7:03 PM, "Joseph L. Casale" wrote: > Searching through the wiki for any indication as to what if any > variables exist > > for the install location in that I can leverage in a script. > > > > Can anyone point me along, I can?t seem to find anything. I want to > place a shell > > script in /opt/freeswitch/scripts that needs a reference to a conf > file that a binary > > it runs is calling. > > > > So now I have in two places hardcoded paths that I was hoping to > avoid, in the dialplan > > and in the shell script. When either of these is run, does there > exist something like > > > > shell_script.sh"/> > > > > and the same for use inside the shell script? > > > > Thanks! > jlc > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:36:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:36:22 -0500 Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf In-Reply-To: <4B30B01B.30809@gmx.net> References: <4B30B01B.30809@gmx.net> Message-ID: Not sure if we have an option to disable info. Even without this, dtmf should go across the bridge fine. Please open up a bug on jira about this Mike On Dec 22, 2009, at 6:40 AM, Peter P GMX wrote: > Hello, > > in a bigger installation with some thousand endpoints in the field we > see, that the endpoint equipment is always using INFO messages > (standard > setting is auto, so the endpoint decides which method to use). I > have 2 > questions to that scenario: > > 1. Is there a way that Freeswitch forces/restricts the endpoint to > use rfc2833 or not to send to allow INFO in the invite message? > 2. Currently INFO messages do not get forwarded from the caller > through freeswitch to called endpoint. How can we enable that FS > is fowarding the INFO messages? > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:38:49 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:38:49 -0500 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> Message-ID: <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins wrote: > > > On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: > Yes, the internal profile exists. > > > > Name > Type Data State > > === > === > === > === > === > === > === > === > === > ====================================================================== > > internal profile > sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) > > internal-ipv6 profile sip:mod_sofia@[:: > 1]:5060 RUNNING (0) > > external profile > sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) > > example.com gateway > sip:joeuser at example.com NOREG > > 192.168.10.25 alias > internal ALIASED > > === > === > === > === > === > === > === > === > === > ====================================================================== > > 3 profiles 1 alias > > > > > I would do a sanity check at this point: put this box and one phone > on a completely separate network with nothing else and see what > happens. > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/31a1d355/attachment.html From mike at jerris.com Tue Dec 22 20:48:36 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:48:36 -0500 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: We expect the g729 sometime very soon, weeks not months away. Mike On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote: > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji > wrote: >> Hello people, >> >> Can someone please clear the following ambiguities with codecs: >> >> Are we definitively able to run pass-through codecs (e.g. G.729) in >> Proxy >> Media mode, or does FS need to be running in bypass-media ? the >> Wiki is not >> clear in this regard > > Yes, you can use proxy media, bypass media, or even regular mode if > you don't transcode (special for g729). Proxy media is really a > special hack that should only be used for T38 passthrough. If you are > using it for other purposes, think about it some more.... > >> When an A-leg has negotiated a pass-through media codec, can the B- >> leg be >> transcoded into a non-pass-through codec, and vice-versa ? think A- >> leg >> incoming with a G.729 codec, and target for B-leg needs to be setup >> with a >> GSM-codec, say > > That would require transcoding - which can't be done if the codec is > pass-through. > >> Where in the developer's set of documentation are codecs >> discussed ? I would >> like to start porting some code of mine for G.729a/b/ab form a ti DSP >> platform to FS. FS lacking full G.729 support is proving quite a >> hindrance, >> and there is no clear direction from the dev community as to when >> the same >> will be available. Incidentally, any news on this effort ? where >> are we with >> code, and what's an ETA for a Beta ? > > I'd say look at the broadvoice or other simple self-contained codecs > are done. Currently the only supported g729 solution is to use a > digium board with mod_dahdi_codec. > > I don't have any info on a software based g729 solution. > >> On the same lines as (3) above, there is a codec dev template in >> the source >> tree. Again, where can I find documentation relating to this ? the >> template >> has hardly any docs at all. >> >> Best regards and warm wishes for a Merry Christmas and a great New >> Year to >> one and all. >> >> Ahmed. >> >> >> -- >> Ahmed A. Ibrahim-Naji Al-Alousi >> Ph.D., MIEE, MBCS >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:52:00 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:52:00 -0500 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Tha being said that does not sound like what you are experiencing Mike On Dec 22, 2009, at 10:29 PM, David Knell wrote: > On the other hand, a u-law WAV turned into L16 and then back to u- > law to > be sent down the line shouldn't suffer any alteration at all - if it > does, the there's something wrong with the translation. > > The quality dropping over time is almost certainly down to something > else. Vinuth -can you get a recording to compare with the original? > > --Dave > > >> If its degrading like that you have bigger issues... the sound >> files played from wav files vs raw PCM files is NO different on a >> land line and I speak from very many years of experience... your >> wav files are ulaw in wav containers thus will never play native >> which might just be part of your problem. You would have to have >> raw headerless data in a .PCMU file for it to play native. >> >> Can you elaborate on your setup a bit more? >> >> /b >> >> On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: >> >>> The audio quality is a lot different when it plays on the >>> landline. And the quality degrades a bit when the message played >>> is lengthy >30s. So I thought it would be better if I have the >>> file in mu-law and play it as is.. >>> >>> Thanks, >>> Vinuth. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From JCasale at activenetwerx.com Tue Dec 22 21:03:15 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 23 Dec 2009 05:03:15 +0000 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> References: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> Message-ID: >For the path in the dialplan I don't think we have any right now but >file a bug on jira and I can try to add them. As for something in the >script itself that is a bit more work but if anyone has a patch to >inject some vars into scripts like that it would be a nice addition. > >Mike Ok, signed up for an account, where does the dialplan part go, FSCORE? Thanks for the help! jlc From marc at kasteris.com Tue Dec 22 21:14:16 2009 From: marc at kasteris.com (Marc Orenberg) Date: Tue, 22 Dec 2009 21:14:16 -0800 (PST) Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking Message-ID: <707756.35973.qm@web50808.mail.re2.yahoo.com> Hello. I've written an application using mod_conference which often has two parties speaking at once and one party listening. When only one party is speaking, the sound quality is fine, but when a second party starts speaking while the first party is still speaking, the second party's voice is cut-off at the beginning, and both parties voices seem to get choppy, like maybe all of the packets aren't getting delivered properly. I'm experiencing this with the latest trunk version (16012). I have the "member-flags" variable set to "waste", and "comfort-noise" is set to "true". I'm not sure where the problem is coming from; I think if it was a VOIP issue I'd hear the same problem when only one party is speaking. Is there something in mod_conference which would try to filter out other voices when one voice is speaking? I'd really appreciate any suggestions about where to look to find this problem. Thanks in advance, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/0071969d/attachment-0001.html From mike at jerris.com Tue Dec 22 21:18:33 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 00:18:33 -0500 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: References: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> Message-ID: <9C787E83-4C76-42F0-85D9-00BE7D80F916@jerris.com> Sounds right to me, just assign it to me if it lets you Mike On Dec 23, 2009, at 12:03 AM, "Joseph L. Casale" wrote: >> For the path in the dialplan I don't think we have any right now but >> file a bug on jira and I can try to add them. As for something in >> the >> script itself that is a bit more work but if anyone has a patch to >> inject some vars into scripts like that it would be a nice addition. >> >> Mike > > Ok, signed up for an account, where does the dialplan part go, FSCORE? > Thanks for the help! > jlc > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From rob4manhere at gmail.com Tue Dec 22 21:23:10 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Tue, 22 Dec 2009 23:23:10 -0600 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <707756.35973.qm@web50808.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> Message-ID: <07801A24-05AA-4E78-9128-04CA349F59E6@gmail.com> Try setting your energy-level down, at 0 for instance. If it helps, then increase until you find a happy medium. On Dec 22, 2009, at 11:14 PM, Marc Orenberg wrote: > Hello. I've written an application using mod_conference which often > has two parties speaking at once and one party listening. > When only one party is speaking, the sound quality is fine, but when > a second party starts speaking while the first party is still > speaking, the second party's > voice is cut-off at the beginning, and both parties voices seem to > get choppy, like maybe all of the packets aren't getting delivered > properly. > > I'm experiencing this with the latest trunk version (16012). > I have the "member-flags" variable set to "waste", and "comfort- > noise" is set to "true". > > I'm not sure where the problem is coming from; I think if it was a > VOIP issue I'd hear the same problem when only one party is speaking. > Is there something in mod_conference which would try to filter out > other voices when one voice is speaking? > > I'd really appreciate any suggestions about where to look to find > this problem. > > Thanks in advance, > Marc > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/42782894/attachment.html From jason at jasonjgw.net Tue Dec 22 21:29:58 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Dec 2009 16:29:58 +1100 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <707756.35973.qm@web50808.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> Message-ID: <20091223052958.GA7600@jdc.jasonjgw.net> Marc Orenberg wrote: > Is there something in mod_conference which would try to filter out other > voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. From marc at kasteris.com Tue Dec 22 21:45:57 2009 From: marc at kasteris.com (Marc Orenberg) Date: Tue, 22 Dec 2009 21:45:57 -0800 (PST) Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <20091223052958.GA7600@jdc.jasonjgw.net> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> Message-ID: <568455.74607.qm@web50801.mail.re2.yahoo.com> Thanks Rob, thanks Jason. I'm going to try this first thing tomorrow. The "energy-level" paramter is described in the file as, "Energy level required for audio to be sent to the other users", so one would think that this would have no effect if member-flags is set to "waste", right? ________________________________ From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 23, 2009 12:29:58 AM Subject: Re: [Freeswitch-users] mod_conference voice problems when two parties speaking Marc Orenberg wrote: > Is there something in mod_conference which would try to filter out other > voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/7fd9939c/attachment.html From dule.maillist at gmail.com Tue Dec 22 22:00:51 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 23 Dec 2009 01:00:51 -0500 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <568455.74607.qm@web50801.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> <568455.74607.qm@web50801.mail.re2.yahoo.com> Message-ID: <914fc92a0912222200y64963743n7a6f82d48cca3247@mail.gmail.com> No, from my understanding that's not how it works. Waste just means it'll always send RTP packets, doesn't mean it will contain audio... so if you have audio that's under your energy threshold, you still won't hear it. Dan On Wed, Dec 23, 2009 at 12:45 AM, Marc Orenberg wrote: > Thanks Rob, thanks Jason. > I'm going to try this first thing tomorrow. > The "energy-level" paramter is described in the file as, "Energy level > required for audio to be sent to the other users", so one would think that > this would have no effect if member-flags is set to "waste", right? > > ------------------------------ > *From:* Jason White > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wed, December 23, 2009 12:29:58 AM > *Subject:* Re: [Freeswitch-users] mod_conference voice problems when two > parties speaking > > Marc Orenberg wrote: > > Is there something in mod_conference which would try to filter out other > > voices when one voice is speaking? > > Try reducing the energy level parameter in case this is the issue. It's > 7/8/9 > on the key pad during the call, or via the conference command, or the > settings > in conference.conf.xml. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/27a5c779/attachment.html From rob4manhere at gmail.com Tue Dec 22 22:03:12 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 23 Dec 2009 00:03:12 -0600 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <568455.74607.qm@web50801.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> <568455.74607.qm@web50801.mail.re2.yahoo.com> Message-ID: <4B4B1517-6473-486A-B726-3454B0B68C3D@gmail.com> No, they are related by different. Waste means mod_conference is *sending* to the recipients all of the time.. whether the conference is silent or not. Energy-level is the hump each channels has to get over before mod_conference accepts audio from that line and includes that channel in the conference. By setting it to 0, it bridges all channels all of the time, whether they are quite or not. Rob On Dec 22, 2009, at 11:45 PM, Marc Orenberg wrote: > Thanks Rob, thanks Jason. > I'm going to try this first thing tomorrow. > The "energy-level" paramter is described in the file as, "Energy > level required for audio to be sent to the other users", so one > would think that this would have no effect if member-flags is set to > "waste", right? > > From: Jason White > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 23, 2009 12:29:58 AM > Subject: Re: [Freeswitch-users] mod_conference voice problems when > two parties speaking > > Marc Orenberg wrote: > > Is there something in mod_conference which would try to filter out > other > > voices when one voice is speaking? > > Try reducing the energy level parameter in case this is the issue. > It's 7/8/9 > on the key pad during the call, or via the conference command, or > the settings > in conference.conf.xml. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/bfc81026/attachment-0001.html From dave at 3c.co.uk Tue Dec 22 22:47:08 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 23 Dec 2009 06:47:08 +0000 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> Message-ID: <1261550828.18473.64.camel@local.freepabx.com> On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: > That being said, ulaw l16 alaw will cause degredation and any other > modifications such as volume adjustment in this path will make it > worse. Indeed. Storing prompts as 8k, 16-bit WAVs makes a lot of sense. [I am inordinately pleased with the above] --Dave From vinuth.madinur at gmail.com Tue Dec 22 23:08:53 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 12:38:53 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> Message-ID: <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> My setup is as follows: FreeSWITCH -> SIP Trunk -> PSTN. >From freeswitch, I'm making outbound calls using event socket via the "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is default settings. Using "playback" application, I'm playing a mu-law audio. I'm also starting the "vmd" application, so that I can replay the message on beep. Thanks for your suggestion on native format. I'll try it. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:41 AM, Brian West wrote: > If its degrading like that you have bigger issues... the sound files played > from wav files vs raw PCM files is NO different on a land line and I speak > from very many years of experience... your wav files are ulaw in wav > containers thus will never play native which might just be part of your > problem. You would have to have raw headerless data in a .PCMU file for it > to play native. > > Can you elaborate on your setup a bit more? > > /b > > On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > > > The audio quality is a lot different when it plays on the landline. And > the quality degrades a bit when the message played is lengthy >30s. So I > thought it would be better if I have the file in mu-law and play it as is.. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9ee6022d/attachment.html From vinuth.madinur at gmail.com Tue Dec 22 23:14:11 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 12:44:11 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261550828.18473.64.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> <1261550828.18473.64.camel@local.freepabx.com> Message-ID: <910309030912222314hd5e6f81o7b0ca0c459d1d542@mail.gmail.com> On Wed, Dec 23, 2009 at 12:17 PM, David Knell wrote: > On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: > > That being said, ulaw l16 alaw will cause degredation and any other > > modifications such as volume adjustment in this path will make it > > worse. > > Indeed. Storing prompts > as 8k, 16-bit WAVs > makes a lot of sense. > > [I am inordinately pleased with the above] > > --Dave > > Thanks, will try and get back. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9a4ae9f3/attachment.html From kristoff.bonne at skypro.be Wed Dec 23 01:54:58 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Wed, 23 Dec 2009 10:54:58 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: References: <4B3142F7.1080600@skypro.be> Message-ID: <4B31E8F2.6040907@skypro.be> Hi Rupa, None. That's exactly the point. Everything has to be done over the usb "HID" interface. I've been reading about HID yesterday. HID is a usb interface that can be used for a large number of things, ranging from keyboard and game-controllers up to "water-cooling and PC-chassis" and point-of-sale or coin changer devices. It also has a telephony-interface: see page 69 to 72 of this document: http://www.usb.org/developers/devclass_docs/HID1_11.pdf This include call-control, on-hook/off-hook detection, DTMF-related things, etc. Now, the question is this: Is there a way to "plug" this all into freeswitch? Cheerio! Kr. Bonne. Rupa Schomaker schreef: > Interesting. It would have to do more than just dialtone/dtmf though. > Need call control, caller id, etc. What do they ship with it as far > as drivers go? > > On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne > wrote: > >> Hi all, >> >> >> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >> device for just 15 euro. This is a device which has on one side a >> USB-connector and on the other side 2 RJ-11 connectors (one FXO and one >> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >> (see here: http://www.tjnet.com/chips/tiger560C.htm) >> >> >> What is interesting on this device is that is uses standard USB >> device-classes that are by default supported by most operating-systems: >> usb-sound and usb-hid. >> >> >> When I connect it to my server (mac mini 3G running debian), the system >> automatically recognises these two classes >> >> [168391.922479] usbcore: registered new interface driver hiddev >> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on >> usb-0001:10:1b.1-1 >> [168391.939548] usbcore: registered new interface driver usbhid >> [168391.943984] usbhid: v2.6:USB HID core driver >> [168392.154596] usbcore: registered new interface driver snd-usb-audio >> >> >> And -behold- when I connect a handset in one of the port, I even get a >> dialtone and I can sent out DTMF-dialtone which are somehow partly >> (But I have no idea what program actually generates this dialtone !!!) >> >> >> >> Now, the question: >> Any idea if / how this can incorperated into freeswitch? Is there a way >> to use this device to connect a phone to freeswitch without having to go >> throu a SIP-client first. >> >> >> >> Cheerio! Kr. Bonne. >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/01e9921c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/01e9921c/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/01e9921c/attachment.bin From codecomplete at free.fr Wed Dec 23 02:59:47 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 02:59:47 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> Message-ID: <26897658.post@talk.nabble.com> mercutioviz wrote: > Additionally, turn on debugging on the console and capture that output. If > you use fs_cli it has debug output turned on by default. Thanks for the tip. I launched fs_cli, typed ""sofia profile internal siptrace on", and then made a call from XLite to the GS phone, with the same issue. I wish I could go through the debug messages in the CLI, but there's so much data that I can't even see the beginning. Is there a way to reduce the amount of information, eg. only displaying the SIP messages, or only displaying the lines that start with [debug] while ignoring those that start with [notice]? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897658.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 23 03:26:56 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 03:26:56 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26897658.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> Message-ID: <26897904.post@talk.nabble.com> I guess I can limit the amount of debug data in the CLI by choosing the right debug level: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP What is the recommended way to debug SIP connections like I'm having? -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897904.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Wed Dec 23 04:49:15 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 23 Dec 2009 20:49:15 +0800 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port Message-ID: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> Hi all, I'm using FS 1.5, doesn't somebody known something about this problem? My scenario is : A(FreeSwitch) B ------INVITE -------> <----100 Tring-------- <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 c=IN IP4 10.36.143.76 <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 -----200 OK ------> response for UPDATE message <---- 200 OK-------- response for INVITE message, with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 --------ACK ---------> The problem is, B changed the rtp port in UPDATE message and "200 OK" response message, but FS didn't do update , so it still send and receive data from port 55066. Is this a bug in FS? Does someone known something about this problem? Any advice is appreciated! -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ff823243/attachment-0001.html From codecomplete at free.fr Wed Dec 23 04:51:35 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 04:51:35 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26897904.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> <26897904.post@talk.nabble.com> Message-ID: <26901641.post@talk.nabble.com> BTW, here's the layout: http://img192.imageshack.us/img192/539/investigatingimmediater.jpg All hosts are located in the same LAN with network address 192.168.0.0/24 and are connected to the hub in the ADSL NAT router. Regardless of whether I use XLite, the GrandStream IP phone, or the analog handset connected to the Linksys 3102, I get the same error: The remote extension (target) rings, but when I pick up the call on the (target) phone, the call is terminated on the target end, and the source extension is immediately redirected to the target extension's voice-mail. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26901641.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yivzhenko at mksat.net Wed Dec 23 05:30:24 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Wed, 23 Dec 2009 15:30:24 +0200 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 In-Reply-To: <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> References: <200912171305.57498.yivzhenko@mksat.net> <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> Message-ID: <200912231530.25380.yivzhenko@mksat.net> Yes i'l be happy to see some working examples :) I can't fully understand how freeswitch conceptually manage presence events. And i not found any information about it in wiki. With default configuration fs sends some notifications to subscribed phones without use any external scripts, but this works very unpredictable. (in most situations BLF change state only once and only for dialing side. after reboot phone BLF state may be change again on some events or may not change any more) if i use the following script blf state changed but only once after reboot the phone #!/usr/bin/php addHeader("proto", "sip"); $e1->addHeader("from", "$Username@$Domain"); $e1->addHeader("login", "$Username@$Domain"); $e1->addHeader("event_type", "presence"); $e1->addHeader("alt_event_type", "dialog"); $e1->addHeader("Presence-Call-Direction", "outbound"); $e1->addHeader("answer-state", $State); $res = $esl1->sendEvent($e1); } SendEvent('220',"mydomain.net","confirmed"); sleep(4); SendEvent('220',"mydomain.net","terminated"); ?> On Tuesday 22 December 2009 21:51:25 mm_202 wrote: > Yuriy, > > The FS wiki has examples of how to control the BLF/MWI using events. > I had no problem getting to work with my GXP2020. > > Let me know if you want some direct code examples. > > -- MM. > > On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko wrote: > > Hallo All! > > I need information about setup BLF on GXP2010/2020 phones with > > Freeswitch. I search in Freeswitch Wiki and maillist archives but find no > > usable information. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From a.alalousi at gmail.com Wed Dec 23 05:46:33 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Wed, 23 Dec 2009 13:46:33 +0000 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: Hi Rupa, Thanks for your feedback. I am currently running proxy mode, but whenever I try to force G.729 on in-bound and out-bound calls, I get an error in my logs to the effect the G.729 is only a pass-through codec. Both originator and reciepient have G.729 codecs. Have you seen this before ? Regards, Ahmed. 2009/12/23 Rupa Schomaker > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji wrote: > > Hello people, > > > > Can someone please clear the following ambiguities with codecs: > > > > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy > > Media mode, or does FS need to be running in bypass-media ? the Wiki is > not > > clear in this regard > > Yes, you can use proxy media, bypass media, or even regular mode if > you don't transcode (special for g729). Proxy media is really a > special hack that should only be used for T38 passthrough. If you are > using it for other purposes, think about it some more.... > > > When an A-leg has negotiated a pass-through media codec, can the B-leg be > > transcoded into a non-pass-through codec, and vice-versa ? think A-leg > > incoming with a G.729 codec, and target for B-leg needs to be setup with > a > > GSM-codec, say > > That would require transcoding - which can't be done if the codec is > pass-through. > > > Where in the developer's set of documentation are codecs discussed ? I > would > > like to start porting some code of mine for G.729a/b/ab form a ti DSP > > platform to FS. FS lacking full G.729 support is proving quite a > hindrance, > > and there is no clear direction from the dev community as to when the > same > > will be available. Incidentally, any news on this effort ? where are we > with > > code, and what's an ETA for a Beta ? > > I'd say look at the broadvoice or other simple self-contained codecs > are done. Currently the only supported g729 solution is to use a > digium board with mod_dahdi_codec. > > I don't have any info on a software based g729 solution. > > > On the same lines as (3) above, there is a codec dev template in the > source > > tree. Again, where can I find documentation relating to this ? the > template > > has hardly any docs at all. > > > > Best regards and warm wishes for a Merry Christmas and a great New Year > to > > one and all. > > > > Ahmed. > > > > > > -- > > Ahmed A. Ibrahim-Naji Al-Alousi > > Ph.D., MIEE, MBCS > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ddfd3f44/attachment.html From a.alalousi at gmail.com Wed Dec 23 05:49:59 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Wed, 23 Dec 2009 13:49:59 +0000 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: Hi Mike, Thanks for this. Is the alpha/dev version in the development repositories somewhere ? I would be very interested to participate. I have some code for the ti dsps which may or may not be useful. Also, I'd very much like to put in the effort to the ti DSPs within FS. Is this something that yourself and other devs would be interested in ? Regards, Ahmed. 2009/12/23 Michael Jerris > We expect the g729 sometime very soon, weeks not months away. > > Mike > > On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote: > > > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji > > wrote: > >> Hello people, > >> > >> Can someone please clear the following ambiguities with codecs: > >> > >> Are we definitively able to run pass-through codecs (e.g. G.729) in > >> Proxy > >> Media mode, or does FS need to be running in bypass-media ? the > >> Wiki is not > >> clear in this regard > > > > Yes, you can use proxy media, bypass media, or even regular mode if > > you don't transcode (special for g729). Proxy media is really a > > special hack that should only be used for T38 passthrough. If you are > > using it for other purposes, think about it some more.... > > > >> When an A-leg has negotiated a pass-through media codec, can the B- > >> leg be > >> transcoded into a non-pass-through codec, and vice-versa ? think A- > >> leg > >> incoming with a G.729 codec, and target for B-leg needs to be setup > >> with a > >> GSM-codec, say > > > > That would require transcoding - which can't be done if the codec is > > pass-through. > > > >> Where in the developer's set of documentation are codecs > >> discussed ? I would > >> like to start porting some code of mine for G.729a/b/ab form a ti DSP > >> platform to FS. FS lacking full G.729 support is proving quite a > >> hindrance, > >> and there is no clear direction from the dev community as to when > >> the same > >> will be available. Incidentally, any news on this effort ? where > >> are we with > >> code, and what's an ETA for a Beta ? > > > > I'd say look at the broadvoice or other simple self-contained codecs > > are done. Currently the only supported g729 solution is to use a > > digium board with mod_dahdi_codec. > > > > I don't have any info on a software based g729 solution. > > > >> On the same lines as (3) above, there is a codec dev template in > >> the source > >> tree. Again, where can I find documentation relating to this ? the > >> template > >> has hardly any docs at all. > >> > >> Best regards and warm wishes for a Merry Christmas and a great New > >> Year to > >> one and all. > >> > >> Ahmed. > >> > >> > >> -- > >> Ahmed A. Ibrahim-Naji Al-Alousi > >> Ph.D., MIEE, MBCS > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > -Rupa > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/fdb0bf11/attachment.html From steveu at coppice.org Wed Dec 23 05:55:00 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 23 Dec 2009 21:55:00 +0800 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <4B322134.7030307@coppice.org> On 12/23/2009 11:29 AM, David Knell wrote: > On the other hand, a u-law WAV turned into L16 and then back to u-law to > be sent down the line shouldn't suffer any alteration at all - if it > does, the there's something wrong with the translation. > > The quality dropping over time is almost certainly down to something > else. Vinuth -can you get a recording to compare with the original? > A linear->ulaw->linear->ulaw->linear->ulaw chain *should* only loose quality on the first cycle. However: - Many ulaw implementation are buggy, because they are based on the same broken Sun code. People are gradually waking up and fixing this. The broken implementations can loose considerable quality on each cycle. - Anywhere people fiddle with gains, you will loose quality on each cycle. Steve From codecomplete at free.fr Wed Dec 23 06:22:28 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 06:22:28 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26901641.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> <26897904.post@talk.nabble.com> <26901641.post@talk.nabble.com> Message-ID: <26902800.post@talk.nabble.com> FWIW, I downloaded and compiled the latest trunk (16041), and am still having this issue. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26902800.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From technical at ttnc.co.uk Wed Dec 23 06:26:40 2009 From: technical at ttnc.co.uk (TTNC - Technical) Date: Wed, 23 Dec 2009 14:26:40 +0000 Subject: [Freeswitch-users] RTP/RTCP media whilst recording Message-ID: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Hi There Our Freeswitch cluster receives inbound calls via a SIP trunk from our supplier. I currently have an issue where when a call is sent to voicemail using session:execute("record"), our supplier will terminate the call with a BYE approximately 30 seconds into the recording. They believe the reason for this is our Freeswitch servers are failing to send any RTP/RTCP media while in the recording stage, and therefor they think the call is dead. Is there a way to force Freeswitch to send RTP packets while in the recording stage that I'm missing? Oh, I'm running pretty much the latest svn truck. Any help appreciated. Thanks Russ From mike at jerris.com Wed Dec 23 06:29:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 09:29:38 -0500 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B31E8F2.6040907@skypro.be> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> Message-ID: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Of course there is a way. Depending on the interface your looking at either a freeswitch endpoiny module or an openzap module. Mike On Dec 23, 2009, at 4:54 AM, Kristoff Bonne wrote: > Hi Rupa, > > > None. That's exactly the point. > Everything has to be done over the usb "HID" interface. > > > I've been reading about HID yesterday. HID is a usb interface that > can be used for a large number of things, ranging from keyboard and > game-controllers up to "water-cooling and PC-chassis" and point-of- > sale or coin changer devices. > > > It also has a telephony-interface: > see page 69 to 72 of this document: http://www.usb.org/developers/devclass_docs/HID1_11.pdf > > This include call-control, on-hook/off-hook detection, DTMF-related > things, etc. > > > Now, the question is this: > Is there a way to "plug" this all into freeswitch? > > > > > Cheerio! Kr. Bonne. > > > Rupa Schomaker schreef: >> >> Interesting. It would have to do more than just dialtone/dtmf >> though. >> Need call control, caller id, etc. What do they ship with it as far >> as drivers go? >> >> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >> wrote: >> >>> Hi all, >>> >>> >>> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >>> device for just 15 euro. This is a device which has on one side a >>> USB-connector and on the other side 2 RJ-11 connectors (one FXO >>> and one >>> FSX). Internally, the device seams to contain a tigerjet 560C >>> chipset. >>> (see here: http://www.tjnet.com/chips/tiger560C.htm) >>> >>> >>> What is interesting on this device is that is uses standard USB >>> device-classes that are by default supported by most operating- >>> systems: >>> usb-sound and usb-hid. >>> >>> >>> When I connect it to my server (mac mini 3G running debian), the >>> system >>> automatically recognises these two classes >>> >>> [168391.922479] usbcore: registered new interface driver hiddev >>> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID >>> 06e6:c31c] on >>> usb-0001:10:1b.1-1 >>> [168391.939548] usbcore: registered new interface driver usbhid >>> [168391.943984] usbhid: v2.6:USB HID core driver >>> [168392.154596] usbcore: registered new interface driver snd-usb- >>> audio >>> >>> >>> And -behold- when I connect a handset in one of the port, I even >>> get a >>> dialtone and I can sent out DTMF-dialtone which are somehow partly >>> (But I have no idea what program actually generates this >>> dialtone !!!) >>> >>> >>> >>> Now, the question: >>> Any idea if / how this can incorperated into freeswitch? Is there >>> a way >>> to use this device to connect a phone to freeswitch without having >>> to go >>> throu a SIP-client first. >>> >>> >>> >>> Cheerio! Kr. Bonne. >>> >>> -- >>> jabber/gtalk: kristoff at krbonne.net >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >>> >> >> >> >> > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/cd3e0c5a/attachment.html From steveu at coppice.org Wed Dec 23 06:31:30 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 23 Dec 2009 22:31:30 +0800 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: <4B3229C2.4080109@coppice.org> On 12/23/2009 04:55 AM, Ahmed Naji wrote: > Hello people, > > Can someone please clear the following ambiguities with codecs: > > 1. Are we definitively able to run pass-through codecs (e.g. G.729) > in Proxy Media mode, or does FS need to be running in > bypass-media ? the Wiki is not clear in this regard > 2. When an A-leg has negotiated a pass-through media codec, can the > B-leg be transcoded into a non-pass-through codec, and > vice-versa ? think A-leg incoming with a G.729 codec, and target > for B-leg needs to be setup with a GSM-codec, say > 3. Where in the developer's set of documentation are codecs > discussed ? I would like to start porting some code of mine for > G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 > support is proving quite a hindrance, and there is no clear > direction from the dev community as to when the same will be > available. Incidentally, any news on this effort ? where are we > with code, and what's an ETA for a Beta ? > 4. On the same lines as (3) above, there is a codec dev template in > the source tree. Again, where can I find documentation relating > to this ? the template has hardly any docs at all. > > Best regards and warm wishes for a Merry Christmas and a great New > Year to one and all. The G.729 codec for FS is in testing, and should be out so. If you really want to implement your own, TI DSP code is unlikely to be a good starting point. I assume that code is fixed point. You really need a floating point codec to get any decent speed on a PC. Pentiums and Athlons lack saturating arithmetic (MMX actually has partial saturating arithmetic, but it isn't much use for anything but image processing), so a fixed point implementation ends up very slow. Steve From mike at jerris.com Wed Dec 23 06:32:26 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 09:32:26 -0500 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port In-Reply-To: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> References: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> Message-ID: <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> There is no such thing as freeswitch 1.5. Have you tried latest svn trunk to see if this behavior is the same? Mike On Dec 23, 2009, at 7:49 AM, Lei Tang wrote: > Hi all, I'm using FS 1.5, doesn't somebody known something about > this problem? > My scenario is : > A(FreeSwitch) B > ------INVITE -------> > <----100 Tring-------- > <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 > c=IN IP4 10.36.143.76 > <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 > c=IN IP4 10.36.143.76 > -----200 OK ------> response for UPDATE message > <---- 200 OK-------- response for INVITE message, with > sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 > --------ACK ---------> > The problem is, B changed the rtp port in UPDATE message and "200 > OK" response message, but FS didn't do update , so it still send and > receive data from port 55066. > Is this a bug in FS? Does someone known something about this > problem? Any advice is appreciated! > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/6f87a4e2/attachment.html From brian at freeswitch.org Wed Dec 23 06:55:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 08:55:12 -0600 Subject: [Freeswitch-users] RTP/RTCP media whilst recording In-Reply-To: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> References: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Message-ID: <481BF889-0446-409E-9A07-DEA61D30BAE2@freeswitch.org> What does pretty much mean to you? Can you give me an exact rev? /b On Dec 23, 2009, at 8:26 AM, TTNC - Technical wrote: > Oh, I'm running pretty much the latest svn truck. From lei.tlfly at gmail.com Wed Dec 23 06:56:14 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 23 Dec 2009 22:56:14 +0800 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port In-Reply-To: <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> References: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> Message-ID: <50c41b4e0912230656p18400250yd5a85d5cb0981753@mail.gmail.com> Thanks Michael, sorry for my mistake, I'm using FS 1.0.5pre9, I'll try the lastest svn trunk. 2009/12/23 Michael Jerris > There is no such thing as freeswitch 1.5. Have you tried latest svn trunk > to see if this behavior is the same? > > Mike > > > On Dec 23, 2009, at 7:49 AM, Lei Tang wrote: > > Hi all, I'm using FS 1.5, doesn't somebody known something about this > problem? > My scenario is : > A(FreeSwitch) B > ------INVITE -------> > <----100 Tring-------- > <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 c=IN > IP4 10.36.143.76 > <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 > 10.36.143.76 > -----200 OK ------> response for UPDATE message > <---- 200 OK-------- response for INVITE message, with sdp > m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 > --------ACK ---------> > The problem is, B changed the rtp port in UPDATE message and "200 OK" > response message, but FS didn't do update , so it still send and receive > data from port 55066. > Is this a bug in FS? Does someone known something about this problem? Any > advice is appreciated! > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/710a1f83/attachment-0001.html From brian at freeswitch.org Wed Dec 23 06:56:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 08:56:53 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> Message-ID: VMD will force a transcode anyway too. /b On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > My setup is as follows: > > FreeSWITCH -> SIP Trunk -> PSTN. > > From freeswitch, I'm making outbound calls using event socket via the "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is default settings. Using "playback" application, I'm playing a mu-law audio. I'm also starting the "vmd" application, so that I can replay the message on beep. > > Thanks for your suggestion on native format. I'll try it. > > Thanks, > Vinuth. From rupa at rupa.com Wed Dec 23 07:01:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Dec 2009 09:01:38 -0600 Subject: [Freeswitch-users] RTP/RTCP media whilst recording In-Reply-To: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> References: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Message-ID: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical wrote: > Hi There > > Our Freeswitch cluster receives inbound calls via a SIP trunk from our supplier. I currently have an issue where when a call is sent to voicemail using session:execute("record"), our supplier will terminate the call with a BYE approximately 30 seconds into the recording. > > They believe the reason for this is our Freeswitch servers are failing to send any RTP/RTCP media while in the recording stage, and therefor they think the call is dead. > > Is there a way to force Freeswitch to send RTP packets while in the recording stage that I'm missing? > > Oh, I'm running pretty much the latest svn truck. > > Any help appreciated. > > Thanks > > Russ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From anthony.minessale at gmail.com Wed Dec 23 07:22:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Dec 2009 09:22:26 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> Message-ID: <191c3a030912230722u7ab4c6bw70dd3e7170470ca2@mail.gmail.com> It's more than highly likely you have some other problem like jitter or a bad network connection. Not many people would be able to tell the difference between the sound of an 8k PCM file and the same file encoded to G711 just by listening to it unless there was a severe problem somewhere. Since you are behind NAT you are even more likely to experience drops etc. Record your files as 8k raw 16 bit PCM to get the best out of the file playback in FS and look elsewhere for your audio issues. You can always make sure you are using the latest build of FS to rule out any temporary issues in the code. On Wed, Dec 23, 2009 at 8:56 AM, Brian West wrote: > VMD will force a transcode anyway too. > > /b > > On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > > > My setup is as follows: > > > > FreeSWITCH -> SIP Trunk -> PSTN. > > > > From freeswitch, I'm making outbound calls using event socket via the > "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is > default settings. Using "playback" application, I'm playing a mu-law audio. > I'm also starting the "vmd" application, so that I can replay the message on > beep. > > > > Thanks for your suggestion on native format. I'll try it. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/5f0a921c/attachment.html From codecomplete at free.fr Wed Dec 23 07:39:23 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 07:39:23 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: <26903707.post@talk.nabble.com> More information: I can dial the default extensions like 9999 just fine. It's only when I call any of the IP phones (1001,1002,1003) that the call is immediately forwarded to the callee's voice-mail when the phone goes off the hook. To only keep the SIP messages in the fs_cli screen, typing "sofia loglevel all 0" followed by "sofia profile internal siptrace on" doesn't do the trick, so am unable to post the whole log yet. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26903707.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at acsol.net Wed Dec 23 07:43:17 2009 From: john at acsol.net (John) Date: Wed, 23 Dec 2009 08:43:17 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B30F0CD.8040703@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> Message-ID: <4B323A95.2070503@acsol.net> Still having this issue. Do separate domains need to be real fully qualified domains, or can they just be added as in Company1, 2, 3, etc? On 12/22/2009 9:16 AM, John wrote: > Thanks Brian. I did have both force-register-domain and > force-register-db-domain commented in both the internal.xml and > internal-ipv6.xml. The phones appear to register to the company1 domain, > as shown in sofia status profile company1; however I have noticed that > when I try to make a call to another a phone in the same domain, the > system is trying to call sofia/internal/1004 at company1 -- this is when we > get the message, user not registered. If I can the phones to just > register to the IP address of the machine, they call fine and is shows > sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan > problem? In both cases I am just using the sample dialplan. > > > > > On 12/22/2009 8:13 AM, Brian West wrote: > >> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. >> >> /b >> >> On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: >> >> >> >>> I have Freeswitch setup and working as a single tenant >>> system mostly using the default configuration. Trying to >>> convert to a multitenant environment, I have used both the >>> Multi-tenant and Multiple Companies wiki's. I get the phone >>> to register, can call out using the external profile to a >>> ITSP, can call music on hold; however I can not call other >>> users in the company. >>> It appears that when logged in with single company and >>> default context it sucessfully calls other internal phones >>> with bridge to >>> "sofia/internal/sip:extersion at public-IP:translated-port"; >>> however when I log into "Company1" with the phones, it tries >>> "sofia/internal/dialed-extension at Company1" ... I also get >>> "User not Registered". The dialplans are the same either >>> way. >>> >>> Any ideas? >>> >>> Thanks >>> John >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mattdfong at gmail.com Wed Dec 23 07:48:02 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 23 Dec 2009 07:48:02 -0800 Subject: [Freeswitch-users] forcing ptime settings Message-ID: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and it seemed to fix the really slow playback problem. but now I'm getting a [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 error and in some sessions the audio is playing back too fast (at 3x the speed). Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/2b56fcd2/attachment.html From brian at freeswitch.org Wed Dec 23 07:48:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 09:48:48 -0600 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B323A95.2070503@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> <4B323A95.2070503@acsol.net> Message-ID: Yes DNS is required for this to work properly. /b On Dec 23, 2009, at 9:43 AM, John wrote: > Still having this issue. Do separate domains need to be real fully > qualified domains, or can they just be added as in Company1, 2, 3, etc? > From brian at freeswitch.org Wed Dec 23 07:55:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 09:55:58 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> Message-ID: <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff. /b On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. > > The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 > > I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and it seemed to fix the really slow playback problem. > > but now I'm getting a > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 > > error and in some sessions the audio is playing back too fast (at 3x the speed). > > Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? > > > --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/2346713c/attachment-0001.html From william.suffill at gmail.com Wed Dec 23 08:06:18 2009 From: william.suffill at gmail.com (William Suffill) Date: Wed, 23 Dec 2009 11:06:18 -0500 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Message-ID: <6b65470d0912230806j4a9c6297r8e393c5c735fe079@mail.gmail.com> I haven't played with any of these usb-to-rj11 in a long time but from what I do recall it is picked up as an audio device. That way a regular telephone can be used for mic/speaker for a softphone. Given that you should be able to get audio to/from it using portaudio for starters. Pretty basic and doesn't take advantage of all the features but it would be at least a place to look. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/98451cc2/attachment.html From rupa at rupa.com Wed Dec 23 08:20:25 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Dec 2009 10:20:25 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> Message-ID: They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes. On Wed, Dec 23, 2009 at 9:55 AM, Brian West wrote: > That usually means they are saying 30 but sending 10 which is broken.. you > can't say hey i'm sending 30 and then send 10... find a new provider or beat > them to death with a cluebat in hopes they fix their broken stuff. > > /b > > On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > > I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm > having some trouble playing .wav files into the media stream using > FreeSWITCH. > > The audio either comes out really slow, or really fast. So a 60 second .wav > file is either finished playing in 90 seconds (really slow) or finishes > playing in 20 seconds (really fast). I believe this is caused by different > ptime values that are being setup in the session. In the FreeSWITCH console > I often received this error > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 20 > > I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and > it seemed to fix the really slow playback problem. > > but now I'm getting a > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 10 > > error and in some sessions the audio is playing back too fast (at 3x the > speed). > > Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the > ptime values? Are there any other work arounds? > > > --matt > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/e98ec9cc/attachment.html From mattdfong at gmail.com Wed Dec 23 08:41:06 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 23 Dec 2009 08:41:06 -0800 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> Message-ID: <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue? --matt On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker wrote: > They don't operate their own voip gateways, just run an SBC in front of a > bunch of other providers. So someone they are reselling is using Sonus > gear. I use them to originate to some destinations but in the US I avoid > them due to the sonus stuff that pops up on certain routes. > > On Wed, Dec 23, 2009 at 9:55 AM, Brian West wrote: > >> That usually means they are saying 30 but sending 10 which is broken.. you >> can't say hey i'm sending 30 and then send 10... find a new provider or beat >> them to death with a cluebat in hopes they fix their broken stuff. >> >> /b >> >> On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: >> >> I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm >> having some trouble playing .wav files into the media stream using >> FreeSWITCH. >> >> The audio either comes out really slow, or really fast. So a 60 second >> .wav file is either finished playing in 90 seconds (really slow) or finishes >> playing in 20 seconds (really fast). I believe this is caused by different >> ptime values that are being setup in the session. In the FreeSWITCH console >> I often received this error >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they >> meant to say was 20 >> >> I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and >> it seemed to fix the really slow playback problem. >> >> but now I'm getting a >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they >> meant to say was 10 >> >> error and in some sessions the audio is playing back too fast (at 3x the >> speed). >> >> Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" >> the ptime values? Are there any other work arounds? >> >> >> --matt >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/7a595d8f/attachment.html From mrene_lists at avgs.ca Wed Dec 23 08:53:25 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 23 Dec 2009 11:53:25 -0500 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> Message-ID: <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 23-Dec-09, at 11:41 AM, Matthew Fong wrote: > If I only care about outbound audio, is there a way to force the > audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? > Or is there still this same issue? > > --matt > > On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker wrote: > They don't operate their own voip gateways, just run an SBC in front > of a bunch of other providers. So someone they are reselling is > using Sonus gear. I use them to originate to some destinations but > in the US I avoid them due to the sonus stuff that pops up on > certain routes. > > On Wed, Dec 23, 2009 at 9:55 AM, Brian West > wrote: > That usually means they are saying 30 but sending 10 which is > broken.. you can't say hey i'm sending 30 and then send 10... find a > new provider or beat them to death with a cluebat in hopes they fix > their broken stuff. > > /b > > On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > >> I use the SIP Termination service from ezcall inc (grnvoip.com) and >> I'm having some trouble playing .wav files into the media stream >> using FreeSWITCH. >> >> The audio either comes out really slow, or really fast. So a 60 >> second .wav file is either finished playing in 90 seconds (really >> slow) or finishes playing in 20 seconds (really fast). I believe >> this is caused by different ptime values that are being setup in >> the session. In the FreeSWITCH console I often received this error >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what >> they meant to say was 20 >> >> I tried forcing the codec and ptime using >> absolute_codec_string='PCMU at 30i' and it seemed to fix the really >> slow playback problem. >> >> but now I'm getting a >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what >> they meant to say was 10 >> >> error and in some sessions the audio is playing back too fast (at >> 3x the speed). >> >> Is there a way I can force ptime to be 30 and avoid FreeSWITCH >> "fixing" the ptime values? Are there any other work arounds? >> >> >> --matt > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ca0ec922/attachment-0001.html From brian at freeswitch.org Wed Dec 23 08:57:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 10:57:26 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> Message-ID: You might also have to set the codec negotiation to scrooge /b On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote: > You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: > > > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9d8e0313/attachment.html From scott.torr.fs at letterboxes.org Wed Dec 23 10:00:43 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 05:00:43 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> Message-ID: <1261591243.5459.1351510407@webmail.messagingengine.com> Yes, I noticed the Jira for the situation where the where the fs controlled skype client generates both an In Band audible DTMF tone and an API signal causing potential confusion for devices down the line. If only the skype client had an option not the generate the tone in the first place that would be good, but then I guess they (skype) think the client would only be an end device ;-) However that is not where I'm having a problem, as I'm purely dealing with 'In band' DTMF tones. The question I had on my mind was did the Skype codec faithfully transport the DTMF tones across the network? http://fs.torr.letterboxes.org/dtmf_compare.html >From these comparisons I would have to say that there in no major filtering or distortion of the DTMF tones when transmitted across the Skype network. So I would have to say that "you can receive calls from skypeIN with inband dtmfs". If someone has a different conclusion please let me know. regards, Scott Torr On Tue, 22 Dec 2009 16:25 +0100, "Giovanni Maruzzelli" wrote: > It is probably because mod_skypiax does not analize incoming audio > looking for dtmf, because the "normal" call from a Skype client peer > sends *both* inband and out of band (signaling) dtmf. > > So, I choose to only detect out of band (signaling) dtmfs, and ignore > possible inband dtmfs (in the audio flow), so to have the most > reliable source (signaling) and spare cpu (not analizing the incoming > audio). > > Never tought you can receive calls from skypeIN with inband dtmfs... > > Open a Jira for this, I'll think about. > > Also, let me know your toughts... > > -giovanni > > > > > On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr > wrote: > > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > > FreeSWITCH Version 1.0.trunk (15787) > > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > > mod_skypiax > > > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > ? > > ? ? > > ? ? > ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > ? ? > > ? > > > > > > > > fs>console loglevel 7 > > > > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > > dtmf tones being detected and decoded by fs in the debug log. > > > > > > If however I use a pstn phone and dial my skypeIN telephone number the > > call comes into fs via skypiax but when I generate dtmf tones on the > > phone they are not detected or decoded by fs. > > > > If I take the record_session file and spectrum analyze the recorded > > tones appear to be within spec. > > > > > > Can anybody suggest why this is not working for me? > > > > > > Is the correct sample rate being used in libteletone_detect.c? > > Does the Goertzel algorithm work for other sample rates other than > > 8000hz? > > > > > > I'm not sure why I can not get this to work? > > > > > > > > regards, > > Scott Torr > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Dec 23 10:07:38 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 23 Dec 2009 19:07:38 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261591243.5459.1351510407@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> <1261591243.5459.1351510407@webmail.messagingengine.com> Message-ID: <7b197bef0912231007r74bc667tbe9c699679afd44@mail.gmail.com> Scott, do as tony wrote, ===== add "start_dtmf" app to your dialplan before bridge to start the inband dtmf detector. ===== -giovanni On Wed, Dec 23, 2009 at 7:00 PM, Scott Torr wrote: > Yes, > I noticed the Jira for the situation where the where the fs controlled > skype client generates both an In Band audible DTMF tone and an API > signal causing potential confusion for devices down the line. If only > the skype client had an option not the generate the tone in the first > place that would be good, but then I guess they (skype) think the client > would only be an end device ;-) > > However that is not where I'm having a problem, as I'm purely dealing > with 'In band' DTMF tones. > > The question I had on my mind was did the Skype codec faithfully > transport the DTMF tones across the network? > > http://fs.torr.letterboxes.org/dtmf_compare.html > > >From these comparisons I would have to say that there in no major > filtering or distortion of the DTMF tones when transmitted across the > Skype network. > > So I would have to say that "you can receive calls from skypeIN with > inband dtmfs". > > > If someone has a different conclusion please let me know. > > regards, > Scott Torr > > > On Tue, 22 Dec 2009 16:25 +0100, "Giovanni Maruzzelli" > wrote: >> It is probably because mod_skypiax does not analize incoming audio >> looking for dtmf, because the "normal" call from a Skype client peer >> sends *both* inband and out of band (signaling) dtmf. >> >> So, I choose to only detect out of band (signaling) dtmfs, and ignore >> possible inband dtmfs (in the audio flow), so to have the most >> reliable source (signaling) and spare cpu (not analizing the incoming >> audio). >> >> Never tought you can receive calls from skypeIN with inband dtmfs... >> >> Open a Jira for this, I'll think about. >> >> Also, let me know your toughts... >> >> -giovanni >> >> >> >> >> On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr >> wrote: >> > ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> > FreeSWITCH Version 1.0.trunk (15787) >> > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> > mod_skypiax >> > >> > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> > >> > >> > ? >> > ? ? >> > ? ?> > ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> > ? ? >> > ? >> > >> > >> > >> > fs>console loglevel 7 >> > >> > >> > If I dial 501 from from a sip phone using "inband" dtmf I can see the >> > dtmf tones being detected and decoded by fs in the debug log. >> > >> > >> > If however I use a pstn phone and dial my skypeIN telephone number the >> > call comes into fs via skypiax but when I generate dtmf tones on the >> > phone they are not detected or decoded by fs. >> > >> > If I take the record_session file and spectrum analyze the recorded >> > tones appear to be within spec. >> > >> > >> > Can anybody suggest why this is not working for me? >> > >> > >> > Is the correct sample rate being used in libteletone_detect.c? >> > Does the Goertzel algorithm work for other sample rates other than >> > 8000hz? >> > >> > >> > I'm not sure why I can not get this to work? >> > >> > >> > >> > regards, >> > Scott Torr >> > >> > >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From scott.torr.fs at letterboxes.org Wed Dec 23 10:08:38 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 05:08:38 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> Message-ID: <1261591718.6380.1351511047@webmail.messagingengine.com> You will need to elaborate a bit more? Not sure where you want me to move the statement to? Also, In what way is a sip call handled differently to a skypiax call? Why would the sip call detect and decode properly? regards, Scott Torr On Tue, 22 Dec 2009 16:26 +0100, "Giovanni Maruzzelli" wrote: > do as anthm say :-) > > On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale > wrote: > > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf > > detector. > > > > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > > wrote: > >> > >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) > >> FreeSWITCH Version 1.0.trunk (15787) > >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > >> mod_skypiax > >> > >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > >> > >> > >> ? > >> ? ? > >> ? ? >> > >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > >> ? ? > >> ? > >> > >> > >> > >> fs>console loglevel 7 > >> > >> > >> If I dial 501 from from a sip phone using "inband" dtmf I can see the > >> dtmf tones being detected and decoded by fs in the debug log. > >> > >> > >> If however I use a pstn phone and dial my skypeIN telephone number the > >> call comes into fs via skypiax but when I generate dtmf tones on the > >> phone they are not detected or decoded by fs. > >> > >> If I take the record_session file and spectrum analyze the recorded > >> tones appear to be within spec. > >> > >> > >> Can anybody suggest why this is not working for me? > >> > >> > >> Is the correct sample rate being used in libteletone_detect.c? > >> Does the Goertzel algorithm work for other sample rates other than > >> 8000hz? > >> > >> > >> I'm not sure why I can not get this to work? > >> > >> > >> > >> regards, > >> Scott Torr > >> > >> > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Dec 23 10:17:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 23 Dec 2009 19:17:21 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261591718.6380.1351511047@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> <1261591718.6380.1351511047@webmail.messagingengine.com> Message-ID: <7b197bef0912231017x2f14892cr60219abda79a1971@mail.gmail.com> Ooops, Had not seen you got it in the dialplan... try to move it after the "answer" and test again. Other than this, only thing that comes in my mind is that the conversion from the pstn to sip (skype partner that gives pstn access) to skype is ruining the dtmfs beyond recognition... but you said that at spectral analisys they're fine... So, I have no idea. -giovanni On Wed, Dec 23, 2009 at 7:08 PM, Scott Torr wrote: > You will need to elaborate a bit more? > > Not sure where you want me to move the /> statement to? > > Also, > In what way is a sip call handled differently to a skypiax call? > Why would the sip call detect and decode properly? > > > ? > ? > ? > ? ? data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > ? > ? > > > > regards, > Scott Torr > > > On Tue, 22 Dec 2009 16:26 +0100, "Giovanni Maruzzelli" > wrote: >> do as anthm say :-) >> >> On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale >> wrote: >> > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf >> > detector. >> > >> > >> > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr >> > wrote: >> >> >> >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> >> FreeSWITCH Version 1.0.trunk (15787) >> >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> >> mod_skypiax >> >> >> >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> >> >> >> >> >> ? >> >> ? ? >> >> ? ?> >> >> >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> >> ? ? >> >> ? >> >> >> >> >> >> >> >> fs>console loglevel 7 >> >> >> >> >> >> If I dial 501 from from a sip phone using "inband" dtmf I can see the >> >> dtmf tones being detected and decoded by fs in the debug log. >> >> >> >> >> >> If however I use a pstn phone and dial my skypeIN telephone number the >> >> call comes into fs via skypiax but when I generate dtmf tones on the >> >> phone they are not detected or decoded by fs. >> >> >> >> If I take the record_session file and spectrum analyze the recorded >> >> tones appear to be within spec. >> >> >> >> >> >> Can anybody suggest why this is not working for me? >> >> >> >> >> >> Is the correct sample rate being used in libteletone_detect.c? >> >> Does the Goertzel algorithm work for other sample rates other than >> >> 8000hz? >> >> >> >> >> >> I'm not sure why I can not get this to work? >> >> >> >> >> >> >> >> regards, >> >> Scott Torr >> >> >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > iax:guest at conference.freeswitch.org/888 >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Wed Dec 23 10:53:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 10:53:25 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: <87f2f3b90912231053l645e841cy8a733b10c4ff1de5@mail.gmail.com> On Tue, Dec 22, 2009 at 7:58 PM, Joseph L. Casale wrote: > >> Am I correct in presuming that Freeswitch will answer a fax from a local > zap based user > >> just like it does from an FXO port connected to a POTS line? What I hope > to do here is > >> catch any call made from that extension (the zap based fax machine/user) > and push its > >> call into the fax module. > > > > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it > gets dialtone > > and dials. Whatever it dials is put into ${destination_number} just like > any SIP phone that > > dials. This extension looks ok. Try it out and let us know how it goes. > > -MC > > Michael, > It worked well, there was however a humorous moment: I was testing with my > own shell script > that simply emailed me directly to my postfix gateway, my exchange server > and mua understood the > uuencoded attachment so once it started working I modified the script to > send to our fax service. > > Well they didn't understand uuencode so the attachment, a single page tiff, > got faxed as 23 pages > of binary :) I used mutt with a redirection to a specific muttrc which > understands mime encoding > which should work everywhere... > > Thanks for the help, you've made an office full of people happy... > Thanks for letting us know that everything worked! I'm glad we didn't have to honor Tony's triple-your-money-back guarantee. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/cb24c678/attachment.html From msc at freeswitch.org Wed Dec 23 11:10:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 11:10:36 -0800 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26903707.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <26903707.post@talk.nabble.com> Message-ID: <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 wrote: > > More information: I can dial the default extensions like 9999 just fine. > It's > only when I call any of the IP phones (1001,1002,1003) that the call is > immediately forwarded to the callee's voice-mail when the phone goes off > the > hook. > > To only keep the SIP messages in the fs_cli screen, typing "sofia loglevel > all 0" followed by "sofia profile internal siptrace on" doesn't do the > trick, so am unable to post the whole log yet. > If you're having this much trouble with the CLI then you might be better off just using a combination of tcpdump and rotating log files. Use this command from the shell to rotate logs: fs_cli -x "fsctl send_sighup" Use the info on this page to collect a pcap: http://wiki.freeswitch.org/wiki/Packet_Capture If you have Wireshark you can open the pcap and do some fun analysis. You can also "follow the tcp stream" and watch the messages going back and forth. Hopefully you'll see what's happening (or not happening) and then we can take it from there. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/15b1124d/attachment.html From larclap at yahoo.com Wed Dec 23 12:48:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 23 Dec 2009 12:48:59 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> Message-ID: <00fb01ca8411$5aa873a0$0ff95ae0$@com> Mike, You were right. I turned iptables off and the phone registered. Thanks so much, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, December 22, 2009 8:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins wrote: On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: Yes, the internal profile exists. Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 192.168.10.25 alias internal ALIASED ============================================================================ ===================== 3 profiles 1 alias I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/6b1c074d/attachment.html From msc at freeswitch.org Wed Dec 23 14:00:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 14:00:01 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call - Holiday Schedule Message-ID: <87f2f3b90912231400q65fefa87g778d1bf4857db142@mail.gmail.com> Hello all! Because the holidays fall on consecutive Fridays this year we decided to have a single conference call on Wednesday Dec 30th at the usual time of 11AM CST. The agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30 Thanks for supporting the weekly calls. Don't forget that we will soon be having giveaways and fun stuff on the calls so be sure to plan on joining us! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/5c47f7e1/attachment.html From larclap at yahoo.com Wed Dec 23 17:49:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 23 Dec 2009 17:49:09 -0800 Subject: [Freeswitch-users] Local call uses public context? Message-ID: <012101ca843b$4921cfd0$db656f70$@com> I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/57f14910/attachment-0001.html From brian at freeswitch.org Wed Dec 23 18:03:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 20:03:28 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <012101ca843b$4921cfd0$db656f70$@com> References: <012101ca843b$4921cfd0$db656f70$@com> Message-ID: <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: > I am trying to setup a second FS box from scratch using v16048. > > What can cause a local call (81002, or 9996) to use context public? It?s a standard vanilla install. > > http://pastebin.freeswitch.org/11629 > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/908b5dd0/attachment.html From kristoff.bonne at skypro.be Thu Dec 24 00:01:04 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Thu, 24 Dec 2009 09:01:04 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Message-ID: <4B331FC0.1050709@skypro.be> Hi Michael, Yesterday I did a quick scan of the complete source-code of both freeswitch and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 (the two possible interface-modules to communicate with HID-devices in linux). So I fear that, at this time, it's not possible to use this device with freeswitch. :-( Cheerio! Kr. Bonne. Michael Jerris schreef: > Of course there is a way. Depending on the interface your looking at > either a freeswitch endpoiny module or an openzap module. > Mike > > On Dec 23, 2009, at 4:54 AM, Kristoff Bonne > wrote: > >> Hi Rupa, >> >> >> None. That's exactly the point. >> Everything has to be done over the usb "HID" interface. >> >> >> I've been reading about HID yesterday. HID is a usb interface that >> can be used for a large number of things, ranging from keyboard and >> game-controllers up to "water-cooling and PC-chassis" and >> point-of-sale or coin changer devices. >> >> >> It also has a telephony-interface: >> see page 69 to 72 of this document: >> http://www.usb.org/developers/devclass_docs/HID1_11.pdf >> >> This include call-control, on-hook/off-hook detection, DTMF-related >> things, etc. >> >> >> Now, the question is this: >> Is there a way to "plug" this all into freeswitch? >> >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Rupa Schomaker schreef: >>> >>> Interesting. It would have to do more than just dialtone/dtmf though. >>> Need call control, caller id, etc. What do they ship with it as far >>> as drivers go? >>> >>> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >>> wrote: >>> >>>> Hi all, >>>> >>>> >>>> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >>>> device for just 15 euro. This is a device which has on one side a >>>> USB-connector and on the other side 2 RJ-11 connectors (one FXO and >>>> one >>>> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >>>> (see here: http://www.tjnet.com/chips/tiger560C.htm) >>>> >>>> >>>> What is interesting on this device is that is uses standard USB >>>> device-classes that are by default supported by most >>>> operating-systems: >>>> usb-sound and usb-hid. >>>> >>>> >>>> When I connect it to my server (mac mini 3G running debian), the >>>> system >>>> automatically recognises these two classes >>>> >>>> [168391.922479] usbcore: registered new interface driver hiddev >>>> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID >>>> 06e6:c31c] on >>>> usb-0001:10:1b.1-1 >>>> [168391.939548] usbcore: registered new interface driver usbhid >>>> [168391.943984] usbhid: v2.6:USB HID core driver >>>> [168392.154596] usbcore: registered new interface driver snd-usb-audio >>>> >>>> >>>> And -behold- when I connect a handset in one of the port, I even get a >>>> dialtone and I can sent out DTMF-dialtone which are somehow partly >>>> (But I have no idea what program actually generates this dialtone !!!) >>>> >>>> >>>> >>>> Now, the question: >>>> Any idea if / how this can incorperated into freeswitch? Is there a >>>> way >>>> to use this device to connect a phone to freeswitch without having >>>> to go >>>> throu a SIP-client first. >>>> >>>> >>>> >>>> Cheerio! Kr. Bonne. >>>> >>>> -- >>>> jabber/gtalk: kristoff at krbonne.net >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> >>> >>> >>> >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment.bin From gmaruzz at celliax.org Thu Dec 24 00:09:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 24 Dec 2009 09:09:44 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B331FC0.1050709@skypro.be> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> <4B331FC0.1050709@skypro.be> Message-ID: <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> Hi Kristoff, if you can send me infos on where to buy it, I probably can adapt my upcoming mod_gsmopen endpoint to manage your device... -giovanni On Thu, Dec 24, 2009 at 9:01 AM, Kristoff Bonne wrote: > Hi Michael, > > > > Yesterday I did a quick scan of the complete source-code of both freeswitch > and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 > (the two possible interface-modules to communicate with HID-devices in > linux). > > So I fear that, at this time, it's not possible to use this device with > freeswitch. :-( > > > > Cheerio! Kr. Bonne. > > > Michael Jerris schreef: > > Of course there is a way.? Depending on the interface your looking at either > a freeswitch endpoiny module or an openzap module. > Mike > > On Dec 23, 2009, at 4:54 AM, Kristoff Bonne > wrote: > > Hi Rupa, > > > None. That's exactly the point. > Everything has to be done over the usb "HID" interface. > > > I've been reading about HID yesterday. HID is a usb interface that can be > used for a large number of things, ranging from keyboard and > game-controllers up to "water-cooling and PC-chassis" and point-of-sale or > coin changer devices. > > > It also has a telephony-interface: > see page 69 to 72 of this document: > http://www.usb.org/developers/devclass_docs/HID1_11.pdf > > This include call-control, on-hook/off-hook detection, DTMF-related things, > etc. > > > Now, the question is this: > Is there a way to "plug" this all into freeswitch? > > > > > Cheerio! Kr. Bonne. > > > Rupa Schomaker schreef: > > Interesting.? It would have to do more than just dialtone/dtmf though. > ?Need call control, caller id, etc.? What do they ship with it as far > as drivers go? > > On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne > wrote: > > Hi all, > > > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) > > > What is interesting on this device is that is uses standard USB > device-classes that are by default supported by most operating-systems: > usb-sound and usb-hid. > > > When I connect it to my server (mac mini 3G running debian), the system > automatically recognises these two classes > > [168391.922479] usbcore: registered new interface driver hiddev > [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on > usb-0001:10:1b.1-1 > [168391.939548] usbcore: registered new interface driver usbhid > [168391.943984] usbhid: v2.6:USB HID core driver > [168392.154596] usbcore: registered new interface driver snd-usb-audio > > > And -behold- when I connect a handset in one of the port, I even get a > dialtone and I can sent out DTMF-dialtone which are somehow partly > (But I have no idea what program actually generates this dialtone !!!) > > > > Now, the question: > Any idea if / how this can incorperated into freeswitch? Is there a way > to use this device to connect a phone to freeswitch without having to go > throu a SIP-client first. > > > > Cheerio! Kr. Bonne. > > -- > jabber/gtalk: kristoff at krbonne.net > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From kristoff.bonne at skypro.be Thu Dec 24 00:22:31 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Thu, 24 Dec 2009 09:22:31 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> <4B331FC0.1050709@skypro.be> <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> Message-ID: <4B3324C7.7090804@skypro.be> Hi Giovanni, I actually got if from my local "carrefour" supermarket (you know, the french supermarket-chain). The product they sell is this: http://www.profoon.nl/TradePoint/Item_View?itemNo=IP-150 It's called a "skype-gateway" (as you can use it with the skype softphone too). Cheerio! Kr. Bonne. Giovanni Maruzzelli schreef: > Hi Kristoff, > > if you can send me infos on where to buy it, I probably can adapt my > upcoming mod_gsmopen endpoint to manage your device... > > -giovanni > > On Thu, Dec 24, 2009 at 9:01 AM, Kristoff Bonne > wrote: > >> Hi Michael, >> >> >> >> Yesterday I did a quick scan of the complete source-code of both freeswitch >> and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 >> (the two possible interface-modules to communicate with HID-devices in >> linux). >> >> So I fear that, at this time, it's not possible to use this device with >> freeswitch. :-( >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Michael Jerris schreef: >> >> Of course there is a way. Depending on the interface your looking at either >> a freeswitch endpoiny module or an openzap module. >> Mike >> >> On Dec 23, 2009, at 4:54 AM, Kristoff Bonne >> wrote: >> >> Hi Rupa, >> >> >> None. That's exactly the point. >> Everything has to be done over the usb "HID" interface. >> >> >> I've been reading about HID yesterday. HID is a usb interface that can be >> used for a large number of things, ranging from keyboard and >> game-controllers up to "water-cooling and PC-chassis" and point-of-sale or >> coin changer devices. >> >> >> It also has a telephony-interface: >> see page 69 to 72 of this document: >> http://www.usb.org/developers/devclass_docs/HID1_11.pdf >> >> This include call-control, on-hook/off-hook detection, DTMF-related things, >> etc. >> >> >> Now, the question is this: >> Is there a way to "plug" this all into freeswitch? >> >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Rupa Schomaker schreef: >> >> Interesting. It would have to do more than just dialtone/dtmf though. >> Need call control, caller id, etc. What do they ship with it as far >> as drivers go? >> >> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >> wrote: >> >> Hi all, >> >> >> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >> device for just 15 euro. This is a device which has on one side a >> USB-connector and on the other side 2 RJ-11 connectors (one FXO and one >> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >> (see here: http://www.tjnet.com/chips/tiger560C.htm) >> >> >> What is interesting on this device is that is uses standard USB >> device-classes that are by default supported by most operating-systems: >> usb-sound and usb-hid. >> >> >> When I connect it to my server (mac mini 3G running debian), the system >> automatically recognises these two classes >> >> [168391.922479] usbcore: registered new interface driver hiddev >> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on >> usb-0001:10:1b.1-1 >> [168391.939548] usbcore: registered new interface driver usbhid >> [168391.943984] usbhid: v2.6:USB HID core driver >> [168392.154596] usbcore: registered new interface driver snd-usb-audio >> >> >> And -behold- when I connect a handset in one of the port, I even get a >> dialtone and I can sent out DTMF-dialtone which are somehow partly >> (But I have no idea what program actually generates this dialtone !!!) >> >> >> >> Now, the question: >> Any idea if / how this can incorperated into freeswitch? Is there a way >> to use this device to connect a phone to freeswitch without having to go >> throu a SIP-client first. >> >> >> >> Cheerio! Kr. Bonne. >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> ________________________________ >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment.bin From mcampbellsmith at gmail.com Thu Dec 24 03:16:31 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 24 Dec 2009 22:16:31 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> Message-ID: <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> Sorry for the delay. Christmas stuff! I have put the log at http://pastebin.freeswitch.org/11632 The SPA3102 has remote IP address 11.11.11.11 (in the log), and internal ip address 192.168.1.3 My local ATA has external IP address 124.11.11.11 (in the log) and internal ip address 192.168.1.120 I have all these set: > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server: stun.freeswitch.org When I set 'Nat Mapping Enable' under tab Line 1 in the SPA3102, I get the following trace. This is all I see and then registration fails. freeswitch at internal> recv 545 bytes from tls/[11.11.11.11]:56886 at 11:12:22.924395: ------------------------------------------------------------------------ REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport From: 2001 ;tag=2f998bc591c1321o0 To: 2001 Call-ID: 115e0ffa-a538d31b at 192.168.1.3 CSeq: 35409 REGISTER Max-Forwards: 70 Contact: 2001 ;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces ------------------------------------------------------------------------ send 678 bytes to tls/[11.11.11.11]:56886 at 11:12:22.972012: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport=56886;received=11.11.11.11 From: 2001 ;tag=2f998bc591c1321o0 To: 2001 ;tag=07c0ymHUv7KXH Call-ID: 115e0ffa-a538d31b at 192.168.1.3 CSeq: 35409 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15490 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="myddns.dydns.org", nonce="35a9849e-f07d-11de-88a5-dbc3ffce4ce8", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ ------------------------------------------------------------------------ Thanks! On Tue, Dec 22, 2009 at 2:03 AM, Brian West wrote: > Can you get me siptraces please. > /b > On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote: > > Thanks Brian and Gad, > > I have stun set and if I do a 'sofia status profile internal', I see > the external IP address of the 3102 ATA, so I assume that stun is > working correctly on the SPA3102. > > These are the options that I have set (according to the 3102 manual). > > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server:?stun.freeswitch.org > > I assume that is all is needed? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From scott.torr.fs at letterboxes.org Thu Dec 24 04:29:10 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 23:29:10 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> Message-ID: <1261657750.2312.1351608571@webmail.messagingengine.com> Hi Anthony, Yes, The "start_dtmf" application is in the dialplan. One question I still have is will the Goertzel algorithm in libteletone_detect.c be able to detect and decode the DTMF tones once they have past through the PSTN and Skype network traversing various codecs? 1) They sound audible and clear. 2) A spectrum graph clearly shows the two frequencies. How bad does the signal need to degrade before the DTMF tones cannot be detected? Can you suggest a way to play recordings through the "start_dtmf" application. This way I can test various wave forms. ** BUG ** Why does samples=0? One thing I have noted is that when "start_ivr_async.c" calls: teletone_dtmf_detect(&pvt->dtmf_detect, frame->data, frame->samples); for a skypiax call the samples=0 for a SIP call the samples=160 I hope this may help track down the problem. Perhaps in time with better understanding of the internal workings of fs and may be able to post solutions rather than problems? regards, Scott Torr On Tue, 22 Dec 2009 09:21 -0600, "Anthony Minessale" wrote: > add "start_dtmf" app to your dialplan before bridge to start the inband > dtmf > detector. > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > wrote: > > > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > > FreeSWITCH Version 1.0.trunk (15787) > > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > > mod_skypiax > > > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > > > > > > > > data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > > > > > > > fs>console loglevel 7 > > > > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > > dtmf tones being detected and decoded by fs in the debug log. > > > > > > If however I use a pstn phone and dial my skypeIN telephone number the > > call comes into fs via skypiax but when I generate dtmf tones on the > > phone they are not detected or decoded by fs. > > > > If I take the record_session file and spectrum analyze the recorded > > tones appear to be within spec. > > > > > > Can anybody suggest why this is not working for me? > > > > > > Is the correct sample rate being used in libteletone_detect.c? > > Does the Goertzel algorithm work for other sample rates other than > > 8000hz? > > > > > > I'm not sure why I can not get this to work? > > > > > > > > regards, > > Scott Torr > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 From nicolas at medularis.com Thu Dec 24 05:05:18 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 10:05:18 -0300 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! Message-ID: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> I just got into the fs cli and when I ran a 'show calls' I got the following message: 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! What does this mean? does it mean the event_socket did not report 8456 events? Why could this happen? The answer to this is pretty critical to me, as I make and monitor calls through the socket. Thanks for your help! Nicolas From yehavi.bourvine at gmail.com Thu Dec 24 06:35:20 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Dec 2009 16:35:20 +0200 Subject: [Freeswitch-users] SNOM shared lines with TLS problems? Message-ID: Hello, Is there anyone who is using SNOM with TLS encryption and shared lines and it works? We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: - With non-shared line UDP and TLS both work ok. - With shared lines UDP works ok. - with shared line TLS works as long as only one phone is registered. - After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. Anyone has this setup working and can share some tips? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/c5fb4a1e/attachment-0001.html From nicolas at medularis.com Thu Dec 24 07:19:45 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 12:19:45 -0300 Subject: [Freeswitch-users] Javascript system calls Message-ID: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> Hi, I wanted to know what is the javascript equivalent of lua's os.execute(). I need to run a command from within a js script. Thanks! Nicolas From nicolas at medularis.com Thu Dec 24 07:37:52 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 12:37:52 -0300 Subject: [Freeswitch-users] Javascript system calls In-Reply-To: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> References: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> Message-ID: <1b46b4e80912240737n56ab173fg70fadb50935397f@mail.gmail.com> I'll reply to myself: the function system() On Thu, Dec 24, 2009 at 12:19 PM, Nicolas Brenner wrote: > Hi, I wanted to know what is the javascript equivalent of lua's > os.execute(). I need to run a command from within a js script. > > Thanks! > > Nicolas > From larclap at yahoo.com Thu Dec 24 08:16:53 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 24 Dec 2009 08:16:53 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> Message-ID: <002901ca84b4$819f9760$84dec620$@com> Brian, Please forgive my slowness, but I'm still having problems with this. When you say that I "really didn't auth the user", did you mean the endpoint/extension? If you did, I upped to svn1 16055 and placed a cidr attribute on the extension and reran the test, resulting in the same output, going to context public. Further, I'm confused about your response about ACL compared with Billy W in an email of 12/22/2009. ".you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from." http://pastebin.freeswitch.org/11633 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 23, 2009 6:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/e23c0dab/attachment.html From msc at freeswitch.org Thu Dec 24 10:59:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Dec 2009 10:59:43 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <002901ca84b4$819f9760$84dec620$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> Message-ID: <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> Lars, Since this question has come up a few times I'm going to write up a nice wiki article on it explaining the differences between letting someone in via an ACL and actually doing digest authentication. In a nutshell, though, it's this: if the user does digest authentication (with the whole REGISTER, 401, REGISTER, 200 OK exchange) then whatever value is in user_context is the context for the calls made by that user. In conf/directory/default/1000.xml (and 1001.xml, etc.) they all have user_context = "default" so when those users register the calls they make are handled in the default context. OTOH, if you let a user in via an ACL they aren't really registered, you've simply opened the door for anyone coming from a particular IP address or IP address range. In that case the calls are handled in the context specified by the context parameter of the sip profile where the calls come in. By default the internal sip profile uses the public context. This is for security reasons. "Paranoid by default" is how you might describe it. You are welcome to change that value to "default" so that calls let in by the ACL are handled just like auth'd calls. Play around with it and let us know how it goes. I think you'll get it once you start modifying settings and making test calls. -MC On Thu, Dec 24, 2009 at 8:16 AM, Lars Zeb wrote: > Brian, > > > > Please forgive my slowness, but I?m still having problems with this. When > you say that I ?really didn?t auth the user?, did you mean the > endpoint/extension? > > > > If you did, I upped to svn1 16055 and placed a cidr attribute on the > extension and reran the test, resulting in the same output, going to context > public. > > > > Further, I?m confused about your response about ACL compared with Billy W > in an email of 12/22/2009. > > > > ??you could simply put these entries in your internal sofia profile. > > > > name="apply-register-acl" value="192.168.0.0/24"/> > > > > In that case, you do not need to include anything in the directory. The > cidr entries in the directory are for providing additional control for each > user id and what IPs they are allowed to make calls from.? > > > > http://pastebin.freeswitch.org/11633 > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > Thanks Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, December 23, 2009 6:03 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Local call uses public context? > > > > 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved > by acl "192.168.10.0/24[]". Access Granted. > > > > Because the context is set on the profile as public... and you really > didn't auth the user so user_context was never set. > > > > /b > > > > On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: > > > > I am trying to setup a second FS box from scratch using v16048. > > > > What can cause a local call (81002, or 9996) to use context public? It?s a > standard vanilla install. > > > > http://pastebin.freeswitch.org/11629 > > > > Thanks, Lars > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/16ccfd9f/attachment-0001.html From larclap at yahoo.com Thu Dec 24 12:31:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 24 Dec 2009 12:31:51 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> Message-ID: <000901ca84d8$1fee8c50$5fcba4f0$@com> Thanks for the reply, Michael. I tried the digest authentication using the cidr and copying the conf/sip_profiles/internal.xml from the distribution, where As a result, one endpoint could not register and another was unauthorized. http://pastebin.freeswitch.org/11634 Then I went changed the context in internal.xml from public to default and And the phones registered OK. So my confusion persists. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 24, 2009 11:00 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? Lars, Since this question has come up a few times I'm going to write up a nice wiki article on it explaining the differences between letting someone in via an ACL and actually doing digest authentication. In a nutshell, though, it's this: if the user does digest authentication (with the whole REGISTER, 401, REGISTER, 200 OK exchange) then whatever value is in user_context is the context for the calls made by that user. In conf/directory/default/1000.xml (and 1001.xml, etc.) they all have user_context = "default" so when those users register the calls they make are handled in the default context. OTOH, if you let a user in via an ACL they aren't really registered, you've simply opened the door for anyone coming from a particular IP address or IP address range. In that case the calls are handled in the context specified by the context parameter of the sip profile where the calls come in. By default the internal sip profile uses the public context. This is for security reasons. "Paranoid by default" is how you might describe it. You are welcome to change that value to "default" so that calls let in by the ACL are handled just like auth'd calls. Play around with it and let us know how it goes. I think you'll get it once you start modifying settings and making test calls. -MC On Thu, Dec 24, 2009 at 8:16 AM, Lars Zeb wrote: Brian, Please forgive my slowness, but I'm still having problems with this. When you say that I "really didn't auth the user", did you mean the endpoint/extension? If you did, I upped to svn1 16055 and placed a cidr attribute on the extension and reran the test, resulting in the same output, going to context public. Further, I'm confused about your response about ACL compared with Billy W in an email of 12/22/2009. ".you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from." http://pastebin.freeswitch.org/11633 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 23, 2009 6:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/bd0f5417/attachment.html From ken at ukgb.net Thu Dec 24 12:42:32 2009 From: ken at ukgb.net (Ken Gillett) Date: Thu, 24 Dec 2009 20:42:32 +0000 Subject: [Freeswitch-users] MacOSX Message-ID: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> Yes, I want to set up FreeSwitch on OSX and at least see how it runs, assuming I can get that far. I've downloaded the latest tarball and run configure which seemed to complete ok. But what next? Do I actually need to run make all install sounds-install moh-install as in some lists of instructions it appears that I just need to run make make install Needless to say I'm not an expert at compiling although I have done a fair bit over the years, just not enough for it to be second nature. So the above apparent ambiguity puzzles me. Also, how can I compile on one machine and then actually run it on a different machine? Is there a relatively simple way to achieve this or must I manually copy all the files to the other machine. What files would that be? Are they all conveniently located in a single folder? Hopeful of some helpful advice, but let's face it, anyone doing this sort of thing on Christmas Eve really ought to get out more:-) Ken G i l l e t t _/_/_/_/_/_/_/_/ From jaugenstine at gmail.com Thu Dec 24 20:40:04 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 24 Dec 2009 20:40:04 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> Message-ID: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Ken, The process is the same for all UNIX like platforms. You run - bootstrap.sh - configure - make - make install Jonathan On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: > Yes, I want to set up FreeSwitch on OSX and at least see how it runs, > assuming I can get that far. > > I've downloaded the latest tarball and run configure which seemed to > complete ok. But what next? Do I actually need to run > > make all install sounds-install moh-install > > as in some lists of instructions it appears that I just need to run > > make > make install > > Needless to say I'm not an expert at compiling although I have done a fair > bit over the years, just not enough for it to be second nature. So the above > apparent ambiguity puzzles me. > > Also, how can I compile on one machine and then actually run it on a > different machine? Is there a relatively simple way to achieve this or must > I manually copy all the files to the other machine. What files would that > be? Are they all conveniently located in a single folder? > > Hopeful of some helpful advice, but let's face it, anyone doing this sort > of thing on Christmas Eve really ought to get out more:-) > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6df46cb2/attachment-0001.html From jason at jasonjgw.net Thu Dec 24 21:23:51 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 25 Dec 2009 16:23:51 +1100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <20091225052351.GA8608@jdc.jasonjgw.net> jonathan augenstine wrote: > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install Unless there are package files (as in .deb and .rpm) for your operating system that can be generated from the sources, in which case you should run the appropriate package building tools. I always prefer to use the package management system, where possible, instead of just compiling and installing software into /usr/local. From ken at ukgb.net Fri Dec 25 01:07:20 2009 From: ken at ukgb.net (Ken Gillett) Date: Fri, 25 Dec 2009 09:07:20 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: What if you've edited modules.conf and why do the install instructions say to run make all install sounds-install moh-install. Also, what is bootstrap.sh? I see it mentioned in some places in the instructions, but not others. Why might I need to run it? On 25 Dec 2009, at 04:40, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan Ken G i l l e t t _/_/_/_/_/_/_/_/ From ken at ukgb.net Fri Dec 25 01:14:31 2009 From: ken at ukgb.net (Ken Gillett) Date: Fri, 25 Dec 2009 09:14:31 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <20091225052351.GA8608@jdc.jasonjgw.net> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <20091225052351.GA8608@jdc.jasonjgw.net> Message-ID: I'd liked to create an install package, but that's a whole new can of worms. Are the FreeSwitch files all installed in a single directory that could be copied to a different machine? On 25 Dec 2009, at 05:23, Jason White wrote: > I always prefer to use the package management system, where possible, instead > of just compiling and installing software into /usr/local. > Ken G i l l e t t _/_/_/_/_/_/_/_/ From qinglan_zeng at hotmail.com Fri Dec 25 01:20:44 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Fri, 25 Dec 2009 09:20:44 +0000 Subject: [Freeswitch-users] GSM modem for Skype In-Reply-To: References: Message-ID: Hello All, I had a GSM VoIP Gateway which can be connected to service providers who use SIP protocal. I remember somebody in Freeswitch community mentioned there is a software which can connect GSM network with Skype network. I'm not sure if this software can work with my GSM VOIP gateway or not. If somebody can send me such software that would be much appriciated and even license fee required would be accepted. I hope I can have a testing on this or I can send this gateway to those guys who have this kind of software for testing. Thanks and Merry Christmas. Daniel Zeng _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/6cf83f27/attachment.html From jason at jasonjgw.net Fri Dec 25 01:43:55 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 25 Dec 2009 20:43:55 +1100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <20091225052351.GA8608@jdc.jasonjgw.net> Message-ID: <20091225094355.GA11565@jdc.jasonjgw.net> Ken Gillett wrote: > I'd liked to create an install package, but that's a whole new can of worms. > > Are the FreeSwitch files all installed in a single directory that could be > copied to a different machine? Yes, but that isn't a substitute for package management. For example, you typically want to preserve configuration files across package updates while having the opportunity to merge changes from newer configuration files. Package management solves this problem; it also solves dependency issues. From lei.tlfly at gmail.com Fri Dec 25 04:36:33 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 25 Dec 2009 20:36:33 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected Message-ID: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> Hi all and merry holiday, I have encounter fs core dump many times when I exit fs_cli, I'm using the fs 1.0.5pre9. I can reproduce this fault by follow steps 1.launch fs with console 2.press ctrl+z to ext from fs console 3.run fs_cli (from local) 4.press ctrl+z to exit fs_cli (or type /bye) 5.fs core dump. This fault is not reproduced every times, but quite frequent. does some have encountered the same problem or have any idea about it? Any suggestion is appreciated! ====here is the back traces in gdb #0 0x02a6593c in ?? () (gdb) where #0 0x02a6593c in ?? () #1 #2 0x0028a410 in __kernel_vsyscall () #3 0x0075870b in write () from /lib/libpthread.so.0 #4 0x0035451a in apr_socket_send (sock=0x85a12f8, buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0x10aaa78) at network_io/unix/sendrecv.c:41 #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) at /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at threadproc/unix/thread.c:138 ====another back trace (gdb) where #0 0x075e893c in ?? () #1 #2 0x002ae410 in __kernel_vsyscall () #3 0x0075870b in write () from /lib/libpthread.so.0 #4 0x001d951a in apr_socket_send (sock=0x92f8650, buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0xe9ea78) at network_io/unix/sendrecv.c:41 #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) at /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at threadproc/unix/thread.c:138 #8 0x007515ab in start_thread () from /lib/libpthread.so.0 #9 0x006a7cfe in clone () from /lib/libc.so.6 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/e74380a9/attachment.html From lei.tlfly at gmail.com Fri Dec 25 04:37:52 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 25 Dec 2009 20:37:52 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected In-Reply-To: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> References: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> Message-ID: <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> BTW my is environment [root at localhost bin]# uname -a Linux localhost.localdomain 2.6.18-164.el5PAE #1 SMP Thu Sep 3 04:10:44 EDT 2009 i686 i686 i386 GNU/Linux [root at localhost bin]# gcc -v ???? specs? ???i386-redhat-linux ????../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk --disable-dssi --enable-plugin --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic --host=i386-redhat-linux ?????posix gcc ?? 4.1.2 20080704 (Red Hat 4.1.2-46) 2009/12/25 Lei Tang > Hi all and merry holiday, I have encounter fs core dump many times when I > exit fs_cli, I'm using the fs 1.0.5pre9. > I can reproduce this fault by follow steps > 1.launch fs with console > 2.press ctrl+z to ext from fs console > 3.run fs_cli (from local) > 4.press ctrl+z to exit fs_cli (or type /bye) > 5.fs core dump. > > This fault is not reproduced every times, but quite frequent. does some > have encountered the same problem or have any idea about it? > Any suggestion is appreciated! > > > ====here is the back traces in gdb > #0 0x02a6593c in ?? () > (gdb) where > #0 0x02a6593c in ?? () > #1 > #2 0x0028a410 in __kernel_vsyscall () > #3 0x0075870b in write () from /lib/libpthread.so.0 > #4 0x0035451a in apr_socket_send (sock=0x85a12f8, > buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0x10aaa78) > at network_io/unix/sendrecv.c:41 > #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, > buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 > #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) > at > /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 > #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at > threadproc/unix/thread.c:138 > > ====another back trace > (gdb) where > #0 0x075e893c in ?? () > #1 > #2 0x002ae410 in __kernel_vsyscall () > #3 0x0075870b in write () from /lib/libpthread.so.0 > #4 0x001d951a in apr_socket_send (sock=0x92f8650, > buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0xe9ea78) > at network_io/unix/sendrecv.c:41 > #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, > buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 > #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) > at > /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 > #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at > threadproc/unix/thread.c:138 > #8 0x007515ab in start_thread () from /lib/libpthread.so.0 > #9 0x006a7cfe in clone () from /lib/libc.so.6 > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/2153e486/attachment-0001.html From jbr at consiglia.dk Fri Dec 25 09:25:23 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 25 Dec 2009 18:25:23 +0100 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: I have added an example on the wiki illustrating how to propagate presence and registrations over a set of networked FSs. Interested? Find it on: http://wiki.freeswitch.org/wiki/Mod_event_multicast. /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/61f23e5b/attachment.html From xengelpublicx at gmail.com Fri Dec 25 10:53:47 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Fri, 25 Dec 2009 21:53:47 +0300 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: <4B350A3B.3060305@gmail.com> On 25.12.2009 20:25, Jon Bruel wrote: > I have added an example on the wiki illustrating how to propagate > presence and registrations over a set of networked FSs. Interested? Yes. From rob4manhere at gmail.com Fri Dec 25 11:41:31 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 25 Dec 2009 13:41:31 -0600 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: <462386EC-8D78-411F-BD30-E42317915041@gmail.com> Very nice- thanks Jon! On Dec 25, 2009, at 11:25 AM, Jon Bruel wrote: > I have added an example on the wiki illustrating how to propagate > presence and registrations over a set of networked FSs. Interested? > Find it on:http://wiki.freeswitch.org/wiki/Mod_event_multicast. /Jon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/6130eb92/attachment.html From neilp at cs.stanford.edu Sat Dec 26 01:29:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 26 Dec 2009 01:29:58 -0800 Subject: [Freeswitch-users] freeswitch init Message-ID: Hi All, On the Freeswitch init page (Debian) the setup you to create a user "freeswitch" to run the FS process. When I did this, freeswitch started up but wasn't able to find/open channels to my sangoma/wanpipe hardware. Is this because: 1. The sangoma hardware was installed and running through root, which is a different user than the one running FS? 2. Wanrouter is not getting started before freeswitch on system boot? Is there a disadvantage to setting up the init script for FS to be run by root? Will that solve this problem? Thanks in advance from a Linux/FS newbie, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/1cc56b24/attachment.html From jason at jasonjgw.net Sat Dec 26 02:05:26 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 26 Dec 2009 21:05:26 +1100 Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <20091226100526.GA13489@jdc.jasonjgw.net> Neil Patel wrote: > On the Freeswitch init page (Debian) the setup you to create a user > "freeswitch" to run the FS process. When I did this, freeswitch started up > but wasn't able to find/open channels to my sangoma/wanpipe hardware. It's likely to be the result of wrong permissions on the device file, or maybe you need to add the freeswitch user to a group that has permission to access the device file. I don't have access to Sangoma hardware, so I don't know what the device files are, but this should be documented somewhere, or perhaps it is noted in your kernel logs. From info at daccii.it Sat Dec 26 02:33:58 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Sat, 26 Dec 2009 11:33:58 +0100 Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <4B35E696.3090408@daccii.it> Hi, you should take a look to /etc/udev/rules.d searching for a zaptel.[something] (extension should be .conf, but i'm not sure). Open it and change user and group in to freeswitch. To change the startup/shutdown sequence, instead, you could use update-rc.d. First look at startup and shutdown order for wanrouter and after execute these commands: sudo update-rc.d -f freeswitch remove sudo update-rc.d freeswitch defaults SS KK On SS put a value greater than wanrouter startup order value and on KK put a value freater than warouter shutdown order value. As "shutdown/startup order value" i refer to SNNxxxxxx or KNNxxxxxx where NN is a numeric value and xxxxxx is the name of the service. You should look for them into /etc/rc3.d and /etc/rc0.d (the first for startup and the second for shutdown) Neil Patel ha scritto: > Hi All, > > On the Freeswitch init page (Debian) the setup you to create a user > "freeswitch" to run the FS process. When I did this, freeswitch > started up but wasn't able to find/open channels to my sangoma/wanpipe > hardware. Is this because: > > 1. The sangoma hardware was installed and running through root, which > is a different user than the one running FS? > 2. Wanrouter is not getting started before freeswitch on system boot? > > Is there a disadvantage to setting up the init script for FS to be run > by root? Will that solve this problem? > > Thanks in advance from a Linux/FS newbie, > Neil -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/95d08164/attachment.vcf From lei.tlfly at gmail.com Sat Dec 26 03:48:57 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Sat, 26 Dec 2009 19:48:57 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected In-Reply-To: <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> References: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> Message-ID: <50c41b4e0912260348k23ed56bdk3f34f9e3113179b3@mail.gmail.com> Hi all, I have found the cause of this problem. It due to some code in a library I loaded into Fs, it set SIGPIPE handler, the handler seemed to be invalid when SIGPIPE is fired, so FS is broken. address, so, 2009/12/25 Lei Tang > BTW my is environment > > [root at localhost bin]# uname -a > Linux localhost.localdomain 2.6.18-164.el5PAE #1 SMP Thu Sep 3 04:10:44 EDT > 2009 i686 i686 i386 GNU/Linux > [root at localhost bin]# gcc -v > ???? specs? > ???i386-redhat-linux > ????../configure --prefix=/usr --mandir=/usr/share/man > --infodir=/usr/share/info --enable-shared --enable-threads=posix > --enable-checking=release --with-system-zlib --enable-__cxa_atexit > --disable-libunwind-exceptions --enable-libgcj-multifile > --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk > --disable-dssi --enable-plugin > --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic > --host=i386-redhat-linux > ?????posix > gcc ?? 4.1.2 20080704 (Red Hat 4.1.2-46) > > > 2009/12/25 Lei Tang > > Hi all and merry holiday, I have encounter fs core dump many times when I >> exit fs_cli, I'm using the fs 1.0.5pre9. >> I can reproduce this fault by follow steps >> 1.launch fs with console >> 2.press ctrl+z to ext from fs console >> 3.run fs_cli (from local) >> 4.press ctrl+z to exit fs_cli (or type /bye) >> 5.fs core dump. >> >> This fault is not reproduced every times, but quite frequent. does some >> have encountered the same problem or have any idea about it? >> Any suggestion is appreciated! >> >> >> ====here is the back traces in gdb >> #0 0x02a6593c in ?? () >> (gdb) where >> #0 0x02a6593c in ?? () >> #1 >> #2 0x0028a410 in __kernel_vsyscall () >> #3 0x0075870b in write () from /lib/libpthread.so.0 >> #4 0x0035451a in apr_socket_send (sock=0x85a12f8, >> buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0x10aaa78) >> at network_io/unix/sendrecv.c:41 >> #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, >> buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 >> #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) >> at >> /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 >> #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at >> threadproc/unix/thread.c:138 >> >> ====another back trace >> (gdb) where >> #0 0x075e893c in ?? () >> #1 >> #2 0x002ae410 in __kernel_vsyscall () >> #3 0x0075870b in write () from /lib/libpthread.so.0 >> #4 0x001d951a in apr_socket_send (sock=0x92f8650, >> buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0xe9ea78) >> at network_io/unix/sendrecv.c:41 >> #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, >> buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 >> #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) >> at >> /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 >> #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at >> threadproc/unix/thread.c:138 >> #8 0x007515ab in start_thread () from /lib/libpthread.so.0 >> #9 0x006a7cfe in clone () from /lib/libc.so.6 >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/5506a45d/attachment-0001.html From max.bridgewater at gmail.com Sat Dec 26 19:51:34 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Sat, 26 Dec 2009 22:51:34 -0500 Subject: [Freeswitch-users] No gsmopen in trunk? Message-ID: Hi, I just did a fresh checkout of Freeswitch from trunk but it seems mod_gsmopen is not there. Am I doing something weird or it has been removed from trunk. Please advise! Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/ccece4dd/attachment.html From yehavi.bourvine at gmail.com Sun Dec 27 03:39:17 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 27 Dec 2009 13:39:17 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> Message-ID: More update: VegaStream engineers found the bug and the fix will be available sometime in January. I am still waiting for AudioCodes... Regards, __Yehavi: 2009/12/17 Mark Campbell-Smith > Thanks Yehavi... > > I posted a question on the Cisco Forum and am waiting a response from > 'engineering' to tell us if they plan to implement standard SRTP > support in the Linksys ATA's. > > TLS is working fine. > > On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine > wrote: > > An interim update: > > > > > > Audiocodes: No success yet. I am working with the manufacturer to debug > it. > > VegaStream: Got the necessary license, configured TLS but it doesn't > work. I > > am working with the local representatives on it. > > > > Regards, __Yehavi: > > > > 2009/12/10 Brian West > >> > >> I have confirmed it works with Polycom, Snom and a few others .... > >> polycom is the hardest to set due to having to put the ca cert into > >> the phone... but other than that its good. > >> > >> /b > >> > >> On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > >> > >> > An intermediate report: > >> > > >> > Audiocodes: TLS works only on outgoing requests, incoming ones are > >> > ignored. I am waiting for Audiocodes' help in order to debug it. > >> > SRTP: worked when no TLS is active. When TLS is active the call is > >> > disconnected when the remote party answers. Still debugging it. > >> > > >> > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > >> > how to enable TLS from the WEB interface. > >> > > >> > Regards, __Yehavi: > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/cdea0c16/attachment.html From kjv at ken-ton.com Sun Dec 27 05:01:03 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Sun, 27 Dec 2009 08:01:03 -0500 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> Message-ID: <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> Setting the codec negotiation to scrooge resolved my problems w/ CallCentric. I'd bet that'd do it for him as well. Lessons Learned by me: 1.) Listen to Brian. 2.) When in doubt, refer to rule 1. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Dec 23, 2009, at 11:57 AM, Brian West wrote: > You might also have to set the codec negotiation to scrooge > > /b > > On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote: > >> You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: >> >> >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/fce4710f/attachment.html From vizentini at hotmail.com Sun Dec 27 09:14:39 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Sun, 27 Dec 2009 17:14:39 +0000 Subject: [Freeswitch-users] SIP registrar Message-ID: Hi,Have you used FS as a registrar server ( handling SIP register/ authorization messages ) with xml_curl for directory lookup?Would FS be suitable for handing registers/authorization messages for about 1,000 U.A.s (with expire 60 s ) ? I am going to make some tests and if you can share your results/experience on this regard it would be very appreciated.I am figuring out if a kamailio/opensips registrar integration with FS is really necessary for a scenario contemplating up to 1K U.A Thanks Paulo _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/4b0d35c8/attachment.html From astmac at stillnewt.org Sun Dec 27 10:52:51 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Sun, 27 Dec 2009 10:52:51 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan Actually, with the release tarballs you don't do bootstrap.sh (unless I am mistaken). I have been building FreeSWITCH on OSX for quite a while (over a year), with good results. I have NOT had any luck building from the SVN, as it seems to throw weird errors on my problems on my platform of choice (PPC OSX Tiger), but the released tarballs seem to work ok (even the pre-release tarballs). I also think that making an OSX package of freeswitch sounds nice, but is a bad idea, UNLESS it's set up in an automated fashion that can stay up to date with changes. Otherwise, lazy OSX people get stuck installing an artifact rather then the best available FreeSWITCH. This happened with Asterisk with the Sunrise telecom people. They ended up creating more problems then good as even years after the fact, silly mac people where still installing the OLD compromised, buggy version just because it was in an OSX installer... Hope this Helps, Marty On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan > > On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: > Yes, I want to set up FreeSwitch on OSX and at least see how it > runs, assuming I can get that far. > > I've downloaded the latest tarball and run configure which seemed to > complete ok. But what next? Do I actually need to run > > make all install sounds-install moh-install > > as in some lists of instructions it appears that I just need to run > > make > make install > > Needless to say I'm not an expert at compiling although I have done > a fair bit over the years, just not enough for it to be second > nature. So the above apparent ambiguity puzzles me. > > Also, how can I compile on one machine and then actually run it on a > different machine? Is there a relatively simple way to achieve this > or must I manually copy all the files to the other machine. What > files would that be? Are they all conveniently located in a single > folder? > > Hopeful of some helpful advice, but let's face it, anyone doing this > sort of thing on Christmas Eve really ought to get out more:-) > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gavin.henry at gmail.com Sun Dec 27 16:03:29 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 28 Dec 2009 00:03:29 +0000 Subject: [Freeswitch-users] SIP registrar In-Reply-To: References: Message-ID: <13ca621c0912271603o7a973ce6kbf74ade6fde79f0c@mail.gmail.com> See the installation guide and example dialplans. It's rather simple to test with SIPp and FS. Thanks. On 27/12/2009, Paulo Vicentini wrote: > > Hi,Have you used FS as a registrar server ( handling SIP register/ > authorization messages ) with xml_curl for directory lookup?Would FS be > suitable for handing registers/authorization messages for about 1,000 U.A.s > (with expire 60 s ) ? > I am going to make some tests and if you can share your results/experience > on this regard it would be very appreciated.I am figuring out if a > kamailio/opensips registrar integration with FS is really necessary for a > scenario contemplating up to 1K U.A Thanks > Paulo > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From sales at cncrepair.com Sun Dec 27 12:22:25 2009 From: sales at cncrepair.com (jim cncrepair.com) Date: Sun, 27 Dec 2009 12:22:25 -0800 Subject: [Freeswitch-users] freeswitch and dahdi give an error: device /dev/zap/channel chan 1 fd 42 (Function not implemented) Message-ID: <4B37C201.3070106@cncrepair.com> I am trying to get freeswitch to talk with a digium tdm400 card. I am using ubuntu and the provided dahdi and freeswitch packages. I have configured /opt/freeswitch/conf/openzap.conf with the following: `--# cat openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 [span zt] name => OpenZAP number => 3 fxo-channel => 3 dahdi's auto generated configuration is the following: `--# cat /etc/dahdi/system.conf # Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 # channel 4, WCTDM/4/3, no module. # Global data loadzone = us defaultzone = us The Freeswitch log is the following: `--# tail -n 900 freeswitch.log| grep zap 2009-12-27 10:05:01.620610 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/modules.conf. 2009-12-27 10:05:01.620738 [NOTICE] zap_io.c:2758 Modules configured: 1 2009-12-27 10:05:01.620765 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/openzap.conf. 2009-12-27 10:05:01.620816 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.621076 [INFO] zap_io.c:2579 Loading IO from /opt/freeswitch/mod/ozmod_zt.so [zt] 2009-12-27 10:05:01.621099 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/zt.conf. 2009-12-27 10:05:01.640363 [INFO] zap_io.c:2379 auto-loaded 'zt' 2009-12-27 10:05:01.651465 [DEBUG] zap_io.c:2400 created span 1 (span1) of type zt 2009-12-27 10:05:01.651607 [DEBUG] zap_io.c:2413 span 1 [name]=[OpenZAP] 2009-12-27 10:05:01.651635 [DEBUG] zap_io.c:2413 span 1 [number]=[1] 2009-12-27 10:05:01.651652 [DEBUG] zap_io.c:2413 span 1 [fxo-channel]=[1] 2009-12-27 10:05:01.651667 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.651847 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 1 fd 42 (Function not implemented) 2009-12-27 10:05:01.651882 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.651968 [DEBUG] zap_io.c:2400 created span 2 (span2) of type zt 2009-12-27 10:05:01.652022 [DEBUG] zap_io.c:2413 span 2 [name]=[OpenZAP] 2009-12-27 10:05:01.652041 [DEBUG] zap_io.c:2413 span 2 [number]=[2] 2009-12-27 10:05:01.652057 [DEBUG] zap_io.c:2413 span 2 [fxo-channel]=[2] 2009-12-27 10:05:01.652071 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.652180 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 2 fd 42 (Function not implemented) 2009-12-27 10:05:01.652207 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.652282 [DEBUG] zap_io.c:2400 created span 3 (span3) of type zt 2009-12-27 10:05:01.652299 [DEBUG] zap_io.c:2413 span 3 [name]=[OpenZAP] 2009-12-27 10:05:01.652315 [DEBUG] zap_io.c:2413 span 3 [number]=[3] 2009-12-27 10:05:01.652344 [DEBUG] zap_io.c:2413 span 3 [fxo-channel]=[3] 2009-12-27 10:05:01.652358 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.652483 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 3 fd 42 (Function not implemented) 2009-12-27 10:05:01.652556 [INFO] zap_io.c:2502 Configured 0 channel(s) 2009-12-27 10:05:01.652573 [ERR] zap_io.c:2765 No modules configured! 2009-12-27 10:05:01.652589 [ERR] mod_openzap.c:2882 Error loading OpenZAP 2009-12-27 10:05:01.652605 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_openzap.so To get to this point. I had to symbolically link /dev/zap to /dev/dahdi and change the permissions of /dev/dahdi. I gave /dev/dhadi 777 permissions. I hope someone can advise us what to do next. Thank you From moises.silva at gmail.com Sun Dec 27 20:23:48 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 27 Dec 2009 23:23:48 -0500 Subject: [Freeswitch-users] freeswitch and dahdi give an error: device /dev/zap/channel chan 1 fd 42 (Function not implemented) In-Reply-To: <4B37C201.3070106@cncrepair.com> References: <4B37C201.3070106@cncrepair.com> Message-ID: On Sun, Dec 27, 2009 at 3:22 PM, jim cncrepair.com wrote: > To get to this point. I had to symbolically link /dev/zap to /dev/dahdi > and change the permissions of /dev/dahdi. I gave /dev/dhadi 777 > permissions. > That's just plain wrong. Remove the symlink. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/7a502030/attachment.html From dome at tel.co.th Sun Dec 27 21:07:00 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 28 Dec 2009 12:07:00 +0700 Subject: [Freeswitch-users] What's problem in SVN ? Message-ID: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> Dear sir, What's problem in SVN ? Not thing update after 23/12/2009 (16055) BG Dome C. From scottferri09 at gmail.com Sun Dec 27 22:44:30 2009 From: scottferri09 at gmail.com (Scott Fernandez) Date: Mon, 28 Dec 2009 12:14:30 +0530 Subject: [Freeswitch-users] Cant able to make call through X-lite In-Reply-To: References: Message-ID: Hello , I have a VOIP account which I configured through X-lite and it works fine. However, when I configure the same account in Freeswitch, the status shows as UP. If I call through freeswitch extension (ex. 1001) via X-lite client it says that USER_BUSY. But when I dial the same number through API command, I am able to make a call and bridge it. What could be the issue and Can any one assist me on this? I have pasted the logs in this URL http://pastebin.freeswitch.org/11635 Regards, Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/2fb9025e/attachment.html From darklion11 at yahoo.com Sun Dec 27 23:49:56 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 27 Dec 2009 23:49:56 -0800 (PST) Subject: [Freeswitch-users] Nibblebill not working if mysql reconnects... Message-ID: <26940540.post@talk.nabble.com> Dear Sir, Nibblebill works when i reinstalled and rebuild freeswitch. But after a while when mysql disconnects and reconnects nibblebill accounts not updating recently. IS there another way to avoid nibblebill for not updating? Thanks, Edmar -- View this message in context: http://old.nabble.com/Nibblebill-not-working-if-mysql-reconnects...-tp26940540p26940540.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Dec 28 00:04:07 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 28 Dec 2009 19:04:07 +1100 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> Message-ID: <20091228080407.GA15032@jdc.jasonjgw.net> Dome Charoenyost wrote: > What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. From gmaruzz at celliax.org Mon Dec 28 00:44:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 28 Dec 2009 09:44:36 +0100 Subject: [Freeswitch-users] No gsmopen in trunk? In-Reply-To: References: Message-ID: <7b197bef0912280044o6a11bd70xe28140f9f0304e14@mail.gmail.com> Ciao Max, I'm working on mod_gsmopen, and I can tell you: mod_gsmopen was never in trunk. Actually has not been released yet. So be patient, in the near future I'll announce here on the mailing list how to download a pre-beta of it. -giovanni On Sun, Dec 27, 2009 at 4:51 AM, Max Bridgewater wrote: > Hi, > > I just did a fresh checkout of Freeswitch from trunk but it seems > mod_gsmopen is not there. Am I doing something weird or it has been removed > from trunk. Please advise! > > Thanks, > Max. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dome at tel.co.th Mon Dec 28 05:15:58 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 28 Dec 2009 20:15:58 +0700 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <20091228080407.GA15032@jdc.jasonjgw.net> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> <20091228080407.GA15032@jdc.jasonjgw.net> Message-ID: <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> Oh...sory.... i forgot chismas and new year. if someone come to thailand please let's me know :) BG Dome C. 2009/12/28 Jason White : > Dome Charoenyost wrote: >> What's problem in SVN ? Not thing update after 23/12/2009 (16055) > > Surely the FreeSWITCH developers are entitled to spend time with their > families/friends after a highly productive year of work. Note that there are > holidays in many countries at this time of year. > > I would like to wish everyone involved in the FreeSWITCH project a pleasant > and refreshing holiday, and much success in 2010. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Dec 28 06:37:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 08:37:43 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> "all" is no longer needed. /b On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > make all install sounds-install moh-install. From brian at freeswitch.org Mon Dec 28 06:46:17 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 08:46:17 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <79116A8B-FCB2-4379-A498-DE53C66C0466@freeswitch.org> You're letting the phones register via the ACL so they never actually do a directory lookup.. domains acl is built from the cidr= attribute on the users in the directory. You have bigger problems if you can't register properly with digest authentication. What does your directory entry look like? /b On Dec 24, 2009, at 2:31 PM, Lars Zeb wrote: > Thanks for the reply, Michael. > > I tried the digest authentication using the cidr and copying the conf/sip_profiles/internal.xml from the distribution, where > > As a result, one endpoint could not register and another was unauthorized. > > http://pastebin.freeswitch.org/11634 > > Then I went changed the context in internal.xml from public to default and > > > And the phones registered OK. So my confusion persists. > > Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/2ecee4e1/attachment-0001.html From brian at freeswitch.org Mon Dec 28 07:25:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 09:25:39 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> Message-ID: <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail. /b On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > This is all I see and then registration fails. From brian at freeswitch.org Mon Dec 28 07:27:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 09:27:15 -0600 Subject: [Freeswitch-users] SNOM shared lines with TLS problems? In-Reply-To: References: Message-ID: <58FEC021-5AA7-484B-B965-D0340ECE968F@freeswitch.org> Shared will require some testing with TLS. I need traces, console logs and you to do some foot work to see if you can provide more details. /b On Dec 24, 2009, at 8:35 AM, Yehavi Bourvine wrote: > Hello, > > Is there anyone who is using SNOM with TLS encryption and shared lines and it works? > > We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: > > With non-shared line UDP and TLS both work ok. > With shared lines UDP works ok. > with shared line TLS works as long as only one phone is registered. > After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. > Anyone has this setup working and can share some tips? > > Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/c150486a/attachment.html From anthony.minessale at gmail.com Mon Dec 28 07:47:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Dec 2009 09:47:20 -0600 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! In-Reply-To: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> References: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> Message-ID: <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> most likely cause would be connecting a socket then not regularly reading from it causing the buffer to fill up. any event socket connection must select on the socket and do regular read attempts or all the events will accumulate on the server side until some sanity check is reached and it begins to throw them away, the fist time there is room in this buffer again (when you consume some from the socket leaving space in the queue) it will report how many have been lost since the last read. One way to cause this would be suspend fs_cli with ctl-z and bring it back to the foreground after some time. On Thu, Dec 24, 2009 at 7:05 AM, Nicolas Brenner wrote: > I just got into the fs cli and when I ran a 'show calls' I got the > following message: > > 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! > > > What does this mean? does it mean the event_socket did not report 8456 > events? Why could this happen? > > The answer to this is pretty critical to me, as I make and monitor > calls through the socket. > > > Thanks for your help! > > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/625d88a5/attachment.html From jerry.richards at teotech.com Mon Dec 28 08:29:40 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 08:29:40 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time In-Reply-To: References: <4B30AB87.3060909@gmx.net> Message-ID: <79F4A32FA1DF436D8BDE8A257B7D3A5A@greyhawk.tonecommander.com> Okay. I uncommented the following lines and the video start works as correctly: Thanks, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time No. The following lines is commented out (internal.xml): Thanks, Jerry -----Original Message----- From: Peter P GMX [mailto:Prometheus001 at gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a > 200 OK with only the PCMU codec. This looks incorrect. The audio > call previously negotiated to the speex/16000 codec, and the re-INVITE > from the caller added the H263-1998 codec. If I re-attempt to start > video at the caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the > complete > scenario: establishing audio call, first failed start video attempt, > and second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From darren at aleph-com.net Mon Dec 28 08:39:03 2009 From: darren at aleph-com.net (Darren Wiebe) Date: Mon, 28 Dec 2009 09:39:03 -0700 Subject: [Freeswitch-users] Billing solutions information In-Reply-To: <1261017871151-4179366.post@n2.nabble.com> References: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> <1261017871151-4179366.post@n2.nabble.com> Message-ID: <4B38DF27.2000700@aleph-com.net> On 12/16/2009 7:44 PM, Amarakeerthi S wrote: > Hi, > > Seems nobody is interested to talk about this topic. I found nibblebill is > great. But doesn't hangup the call when balance goes to zero. The other > problem It allows user to call without checking the balance of the cash > database. Is this natural?. If this works fine we can easily integrate with > a payment gateway like 2checkout. > > Thank you > > > > Lon Baker wrote: > >> Hey everyone, >> >> I am researching billing solutions for Freeswitch and want to consolidate >> the information with what others have found, then add it to the Wiki. >> >> There are seems to be a number of billing solutions by commercial >> providers, >> claiming they can integrate with Freeswitch, but nothing concrete >> explaining >> how far they go. >> >> Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, >> etc? >> >> Mod_nibblebill handles the basics of updating a database table. >> >> The A2Billing teams says they are planning on adding support for >> Freeswitch >> in a few months. >> >> ASTPP.org says they support Freeswitch, but the site hasn't been updated >> since 2008. >> >> If you know about any solutions, links to solutions or any information can >> you send it to me? I will organize it and add it to the wiki. >> >> Thanks! >> >> Lon >> >> _______________________________________________ >> The ASTPP site shouldn't show that it hasn't been updated since 2008 as we've been working on it whenever there is time this year. It "works" but needs some more testing and optimizing to be able to handle higher traffic loads. -- Darren Wiebe Aleph Communications Innovative Data& Voice Solutions Email: darren at aleph-com.net Tel: 1-780-701-7267 Fax: 1-866-274-4506 From nicolas at medularis.com Mon Dec 28 09:28:21 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 28 Dec 2009 14:28:21 -0300 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! In-Reply-To: <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> References: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> Message-ID: <1b46b4e80912280928q30357fb4h27d072c4e8d6f32c@mail.gmail.com> Anthony, thank you very much for your response. The daemon that was reading the events froze, so apparently that was the source of the problem and your explanation fits perfectly. On Mon, Dec 28, 2009 at 12:47 PM, Anthony Minessale wrote: > most likely cause would be connecting a socket then not regularly reading > from it causing the buffer to fill up. > any event socket connection must select on the socket and do regular read > attempts or all the events will accumulate on the server side until some > sanity check is reached and it begins to throw them away, the fist time > there is room in this buffer again (when you consume some from the socket > leaving space in the queue) it will report how many have been lost since the > last read. > > One way to cause this would be suspend fs_cli with ctl-z and bring it back > to the foreground after some time. From mike at jerris.com Mon Dec 28 10:08:04 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Dec 2009 13:08:04 -0500 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> <20091228080407.GA15032@jdc.jasonjgw.net> <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> Message-ID: <6D02BF5B-684B-43C0-92C7-E795A5632124@jerris.com> The issues you ran into are probably sorted out now. Give it a try and if its still not working, post the build errors. Mike On Dec 28, 2009, at 8:15 AM, Dome Charoenyost wrote: > Oh...sory.... i forgot chismas and new year. > if someone come to thailand please let's me know :) > > BG > Dome C. > > > 2009/12/28 Jason White : >> Dome Charoenyost wrote: >>> What's problem in SVN ? Not thing update after 23/12/2009 (16055) >> >> Surely the FreeSWITCH developers are entitled to spend time with their >> families/friends after a highly productive year of work. Note that there are >> holidays in many countries at this time of year. >> >> I would like to wish everyone involved in the FreeSWITCH project a pleasant >> and refreshing holiday, and much success in 2010. >> From jcasale at activenetwerx.com Mon Dec 28 11:54:14 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 28 Dec 2009 19:54:14 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >This is a total work in progress that has not even merged into tree. ?So it is not "known" >to work or not work anywhere. ?Patches to correct issues are welcome. Mike, I took another look at this and don't really know enough about rpm building to diagnose this. Frankly, the format of the latest spec is so wildly different from anything I have ever touched I am at a loss:) Is there a simple manual way for me to properly get the sounds for MOH etc installed? is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location pointed to my /opt/freeswitch/sounds directory? Thanks! jlc From jerry.richards at teotech.com Mon Dec 28 12:21:43 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 12:21:43 -0800 Subject: [Freeswitch-users] Presence Change Distribution Message-ID: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry From mike at jerris.com Mon Dec 28 12:47:47 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Dec 2009 15:47:47 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: the build system already has targets for all of this and there are tarballs you can manually download and extract as well that are located in http://files.freeswitch.org/. if you NEED packages, you will have to wait until that work is complete or figure out what the error is. Mike On Dec 28, 2009, at 2:54 PM, Joseph L. Casale wrote: >> This is a total work in progress that has not even merged into tree. So it is not "known" >> to work or not work anywhere. Patches to correct issues are welcome. > > > Mike, > I took another look at this and don't really know enough about rpm building > to diagnose this. Frankly, the format of the latest spec is so wildly different > from anything I have ever touched I am at a loss:) > > Is there a simple manual way for me to properly get the sounds for MOH etc installed? > is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location > pointed to my /opt/freeswitch/sounds directory? > > Thanks! > jlc > From brian at freeswitch.org Mon Dec 28 13:05:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 15:05:21 -0600 Subject: [Freeswitch-users] twitter.com/freeswitch (its not ours) Message-ID: Dear FreeSWITCHers, Someone has registered the freeswitch name and is squatting on twitter with it. They haven't used it in over a year and I would like to have this for our project as its clearly confusing. If you own this account please contact me off list. Thanks, Brian From msc at freeswitch.org Mon Dec 28 14:25:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 14:25:21 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <87f2f3b90912281425n2b52579ak5b39d5ac4e97e1ad@mail.gmail.com> On Thu, Dec 24, 2009 at 12:31 PM, Lars Zeb wrote: > Thanks for the reply, Michael. > > > > I tried the digest authentication using the cidr and copying the > conf/sip_profiles/internal.xml from the distribution, where > > > > As a result, one endpoint could not register and another was unauthorized. > > > > http://pastebin.freeswitch.org/11634 > > > > Then I went changed the context in internal.xml from public to default and > > > > > > > And the phones registered OK. So my confusion persists. > > Like Brian said in his post, if you let someone in via ACL then there is no directory lookup which means the call is essentially from an anonymous/unidentified party and thus the reason for having the context set to "public" even on the internal profile. This is one of the topics that I intend to cover in the wiki article. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/cb863847/attachment.html From brian at freeswitch.org Mon Dec 28 14:57:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 16:57:49 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <7BAD6694-4FD6-4FEC-A3DF-78BE03129777@freeswitch.org> acl.conf.xml sofia profile: and Then here is an example of a user: Now save that.. restart freeswitch and you now let that user in from 1.2.3.4/32 and set the user_context to default. /b From jerry.richards at teotech.com Mon Dec 28 15:19:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 15:19:41 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail Message-ID: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry From anthony.minessale at gmail.com Mon Dec 28 15:30:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Dec 2009 17:30:44 -0600 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail In-Reply-To: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> Message-ID: <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: > Hello All, > > I posted a FS log into the Pastebin at > http://pastebin.freeswitch.org/11644. > > I am still having the problem where a PSTN-to-Internal call via a Sangoma > A101D card stops ringing the internal phone after about 10 seconds. It > should be ringing for 30 seconds and then go to Voice Mail (as an > Internal-to-Internal call does). > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Tuesday, December 22, 2009 8:02 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail > > > I have a Freeswitch PBX server with an installed Sangoma A101D card > connected to a PRI. Most everything works okay, however when I get an > inbound call from the PSTN, if the call is not answered within about 12 > seconds, the call ends (so it doesn't go to voice mail). If I make a call > from one internal phone to another, then it will go to voice mail after 30 > seconds. How can I get the external call to route to voice mail after 30 > seconds? > > I put a new 11595 log into the pastebin. Do you know any Freeswitch > setting > that might cause this? > > If this issue has been addressed before, what string should I use to search > for it, because I can't find it. > > Thanks, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/602e70a4/attachment-0001.html From help at pdscc.com Mon Dec 28 15:38:38 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 15:38:38 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, Message-ID: <20091228233838.75E611694@sinclaire.sibble.net> Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone client, however I am not seeing the enrollment option popup in zfone 0.92 build 218 on windows in front of an x-lite client. Any suggestions on what I should look at to troubleshoot this? I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but until then.... On 14 Dec 2009 at 20:50, Brian West wrote: > if you don't have ZRTP compiled in as per the wiki it won't work... > their are a few changes coming to this code soon. > > /b > > On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote: > > > Hmm, I emailed the zfoneproject folks about an hour ago asking about a > > release date for zfone3 and was surprised about a half hour later > > with a call > > from PRZ himself. > > > > Here's what I got from the call > > > > 1) the currently released version of zfone already has support for > > secure pbx > > enrollment > > -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From brian at freeswitch.org Mon Dec 28 15:49:19 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 17:49:19 -0600 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <20091228233838.75E611694@sinclaire.sibble.net> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, <20091228233838.75E611694@sinclaire.sibble.net> Message-ID: I'm still not done with this I think we found a bug in the lib... Viktor fixed it today and I'm going to retry after I get done testing G729 more today! ;) /b On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone > client, however I am not seeing the enrollment option popup in zfone 0.92 > build 218 on windows in front of an x-lite client. > > Any suggestions on what I should look at to troubleshoot this? > > I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but > until then.... From msc at freeswitch.org Mon Dec 28 16:07:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 16:07:26 -0800 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> Message-ID: <87f2f3b90912281607w310d8410t44170cf6f20824e0@mail.gmail.com> On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling wrote: > Setting the codec negotiation to scrooge resolved my problems w/ > CallCentric. > > I'd bet that'd do it for him as well. > > *Lessons Learned by me:* > 1.) Listen to Brian. > 2.) When in doubt, refer to rule 1. > Can I get that framed? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/7aaae93b/attachment.html From msc at freeswitch.org Mon Dec 28 16:09:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 16:09:14 -0800 Subject: [Freeswitch-users] Cant able to make call through X-lite In-Reply-To: References: Message-ID: <87f2f3b90912281609r62b9e25du2bb1783a9acfbb15@mail.gmail.com> On Sun, Dec 27, 2009 at 10:44 PM, Scott Fernandez wrote: > Hello , > > I have a VOIP account which I configured through X-lite and it works fine. > However, when I configure the same account in Freeswitch, the status shows > as UP. If I call through freeswitch extension (ex. 1001) via X-lite client > it says that USER_BUSY. But when I dial the same number through API command, > I am able to make a call and bridge it. > > What could be the issue and Can any one assist me on this? > > I have pasted the logs in this URL http://pastebin.freeswitch.org/11635 > > This pb no more... can you re-post? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/8e56859b/attachment.html From help at pdscc.com Mon Dec 28 16:39:26 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 16:39:26 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091228233838.75E611694@sinclaire.sibble.net>, Message-ID: <20091229003926.072E513F5@sinclaire.sibble.net> Coolio, well if you need me to test something, just holler. here's what I'm running FreeSWITCH Version 1.0.trunk (16066) on Ubuntu 9.0.4 zfone 0.92 build 218 (windows xp) with ekiga and x-lite clients tiviphone 2.0.7 (beta) for symbian hopefully will have the tiviphone 2.0.7 (beta) for windows mobile shortly On 28 Dec 2009 at 17:49, Brian West wrote: > I'm still not done with this I think we found a bug in the lib... Viktor > fixed it today and I'm going to retry after I get done testing G729 more > today! ;) > > /b > > On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: > > > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn > trunk, > > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone > > client, however I am not seeing the enrollment option popup in zfone 0.92 > > build 218 on windows in front of an x-lite client. > > > > Any suggestions on what I should look at to troubleshoot this? -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From help at pdscc.com Mon Dec 28 21:21:06 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 21:21:06 -0800 Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch Message-ID: <20091229052106.2592813F5@sinclaire.sibble.net> I am following the wiki page here http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation setup is on Ubuntu 9.0.4 using the debian method, when I run module-assistant build zaptel-source the compilation fails as below, not sure what I am missing, reading further FS wiki pages and googling haven't enlightened me, any suggestions? I am trying to get 2x X100P's working dh_testdir dh_testroot rm -f *-stamp # Delete the generated bristuff symlinks: rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch] # Add here commands to clean up after the build process. rm -rf modexamples rm -f tonezones.txt rm -f version.h rm -rf debian/fake # * Makefile does not exist when running svn-buildpackage # as the source tree is not there. # FIXME: This will fail with an ugly warning on the clean of the # modules build. However only fter the actuual clean. [ ! -f Makefile ] || /usr/bin/make dist-clean || true make[1]: Entering directory `/usr/src/modules/zaptel' make: Entering an unknown directory make: Leaving an unknown directory rm -f torisatool rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest ztscan rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo /usr/bin/make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor make[2]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' CLEAN /usr/src/modules/zaptel/kernel CLEAN /usr/src/modules/zaptel/kernel/.tmp_versions make[2]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[2]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils' rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8 lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8 make[2]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils' make: Entering an unknown directory make: Leaving an unknown directory make[1]: Leaving directory `/usr/src/modules/zaptel' #rm -f debian/manpage.links debian/manpage.refs debian/*.8 dh_clean /usr/bin/make -f debian/rules kdist_clean kdist_config binary-modules make[1]: Entering directory `/usr/src/modules/zaptel' dh_testdir dh_testroot rm -f *-stamp # Delete the generated bristuff symlinks: rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch] # Add here commands to clean up after the build process. rm -rf modexamples rm -f tonezones.txt rm -f version.h rm -rf debian/fake # * Makefile does not exist when running svn-buildpackage # as the source tree is not there. # FIXME: This will fail with an ugly warning on the clean of the # modules build. However only fter the actuual clean. [ ! -f Makefile ] || /usr/bin/make dist-clean || true make[2]: Entering directory `/usr/src/modules/zaptel' make: Entering an unknown directory make: *** menuselect: No such file or directory. Stop. make: Leaving an unknown directory make[2]: [clean] Error 2 (ignored) rm -f torisatool rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest ztscan rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo /usr/bin/make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor make[3]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' make[3]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[3]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils' rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8 lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8 make[3]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils' make: Entering an unknown directory make: *** ppp: No such file or directory. Stop. make: Leaving an unknown directory make[2]: *** [clean] Error 2 make[2]: Leaving directory `/usr/src/modules/zaptel' #rm -f debian/manpage.links debian/manpage.refs debian/*.8 dh_clean for templ in ; do \ cp $templ `echo $templ | sed -e 's/_KVERS_/2.6.28-11-server/g'` ; \ done for templ in `ls debian/*.modules.in` ; do \ test -e ${templ%.modules.in}.backup || cp ${templ%.modules.in} ${templ%.modules.in}.backup 2>/dev/null || true; \ sed -e 's/##KVERS##/2.6.28-11-server/g ;s/#KVERS#/2.6.28-11-server/g ; s/_KVERS_/2.6.28-11-server/g ; s/##KDREV##/2.6.28-11.42 done dh_testdir dh_testroot dh_clean -k cp -a /usr/src/modules/zaptel/debian/generated/* . ./configure checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/bash checking for ln... /bin/ln checking for wget... /usr/bin/wget checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... no checking for usb_init in -lusb... no configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts config.status: creating build_tools/make_firmware_object configure: *** Zaptel build successfully configured *** make MODULES_EXTRA="ds1x1f opvxa1200 wcopenpci cwain qozap zaphfc ztgsm" SUBDIRS_EXTRA="vzaphfc oslec" modules KERNEL_SOURCES=/usr make[2]: Entering directory `/usr/src/modules/zaptel' make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor2.o toris make[3]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' gcc -o /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/makefw.c /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/pciradio.rbt radfw > /usr/src/modules/zaptel/kernel/radfw.h Loaded 42096 bytes from file CC [M] /usr/src/modules/zaptel/kernel/pciradio.o /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/tormenta2.rbt tor2fw > /usr/src/modules/zaptel/kernel/tor2fw. Loaded 69900 bytes from file CC [M] /usr/src/modules/zaptel/kernel/tor2.o CC [M] /usr/src/modules/zaptel/kernel/torisa.o CC [M] /usr/src/modules/zaptel/kernel/wcfxo.o CC [M] /usr/src/modules/zaptel/kernel/wct1xxp.o CC [M] /usr/src/modules/zaptel/kernel/wctdm.o /usr/src/modules/zaptel/kernel/wctdm.c: In function 'wctdm_proslic_getreg_indirect': /usr/src/modules/zaptel/kernel/wctdm.c:671: warning: format not a string literal and no format arguments CC [M] /usr/src/modules/zaptel/kernel/wcte11xp.o CC [M] /usr/src/modules/zaptel/kernel/wcusb.o CC [M] /usr/src/modules/zaptel/kernel/zaptel-base.o /usr/src/modules/zaptel/kernel/zaptel-base.c: In function 'zt_ppp_xmit': /usr/src/modules/zaptel/kernel/zaptel-base.c:1751: warning: comparison of distinct pointer types lacks a cast /usr/src/modules/zaptel/kernel/zaptel-base.c:1814: warning: comparison of distinct pointer types lacks a cast LD [M] /usr/src/modules/zaptel/kernel/zaptel.o CC [M] /usr/src/modules/zaptel/kernel/ztd-eth.o CC [M] /usr/src/modules/zaptel/kernel/ztd-loc.o CC [M] /usr/src/modules/zaptel/kernel/ztdummy.o /usr/src/modules/zaptel/kernel/ztdummy.c: In function 'ztdummy_hr_int': /usr/src/modules/zaptel/kernel/ztdummy.c:203: error: 'struct hrtimer' has no member named 'expires' make[4]: *** [/usr/src/modules/zaptel/kernel/ztdummy.o] Error 1 make[3]: *** [_module_/usr/src/modules/zaptel/kernel] Error 2 make[3]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[2]: *** [modules] Error 2 make[2]: Leaving directory `/usr/src/modules/zaptel' make[1]: *** [binary-modules] Error 2 make[1]: Leaving directory `/usr/src/modules/zaptel' make: *** [kdist_build] Error 2 -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From sharad at coraltele.com Mon Dec 28 22:00:43 2009 From: sharad at coraltele.com (Sharad) Date: Mon, 28 Dec 2009 22:00:43 -0800 (PST) Subject: [Freeswitch-users] Personal Greeting Message-ID: <1262066443847-4226681.post@n2.nabble.com> Hi I am new to Freeswitch so my question may be a stupid question. I just want to know how to disable the personal greeting to the default one. One user has recorded his personal greeting now how can he make this default. I could not find any option for the same. Plz advice. Thanks & regards Sharad garg -- View this message in context: http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sharad at coraltele.com Mon Dec 28 22:23:20 2009 From: sharad at coraltele.com (Sharad) Date: Mon, 28 Dec 2009 22:23:20 -0800 (PST) Subject: [Freeswitch-users] Message Wait Lamp for an unregistered user Message-ID: <1262067800887-4226726.post@n2.nabble.com> Hi, I am trying to integrate Freeswitch as a Media Server with our own SIP Server. For this, initially, I am using freeswitch for Auto Attendant & Voicemail Application. In this case, all the SIP users are registered with my SIP Server & whenever Auto Att or Voicemail application is required, my SIP Server just forward the call to Freeswitch. Everything seems ok & working fine except some small small points. One of the point is - whenever there is a voice message in the mailbox of a user, Freeswitch is not generating the MWI (Notify) to my SIP Server. So just want to know , is there any way so that freeswitch can light up / light -off the message wait lamp for the users of my SIP Server. Thanks in advance for your answers. Regards Sharad -- View this message in context: http://n2.nabble.com/Message-Wait-Lamp-for-an-unregistered-user-tp4226726p4226726.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Mon Dec 28 23:14:51 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 15:14:51 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 Message-ID: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. ====sip trace for trunk 16055 ====re-invite request sent to fs when client hold the line INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: > Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =====bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ======sip trace for fs 1.0.4 =====re-invite request sent to FS when client want to hold the all INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: > Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 Contact: > From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 ACK Date: Tue, 29 Dec 2009 03:20:15 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport Content-Length: 0 INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 INVITE Contact: > Content-Type: application/sdp Content-Length: 473 Date: Tue, 29 Dec 2009 03:20:18 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport v=0 o=sipX 5 35 IN IP4 10.56.90.223 s=call c=IN IP4 10.56.90.223 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE User-Agent: PowerIVR Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE Contact: User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 242 v=0 o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 Contact: > From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 ACK Date: Tue, 29 Dec 2009 03:20:19 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport Content-Length: 0 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/4e5eb8a2/attachment.html From siniypin at gmail.com Tue Dec 29 00:02:24 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 29 Dec 2009 11:02:24 +0300 Subject: [Freeswitch-users] Presence Change Distribution In-Reply-To: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> Message-ID: <2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com> You can try to reduce your registration time. I for one made my client apps send PUBLISH message every minute in addition to reduced registration time. Regards, Robert. 2009/12/28 Jerry Richards > Is there a setting to control how fast FS distributes presence changes to > subscribers? Currently, it appears to take several minutes before I see > presence changes. I would like to see them almost instantaneously, if > possible. > > Thanks and Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/94caa383/attachment.html From Russell.Mosemann at cune.org Tue Dec 29 04:14:38 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Tue, 29 Dec 2009 06:14:38 -0600 Subject: [Freeswitch-users] problems getting openzap compiled for use withfreeswitch In-Reply-To: <20091229052106.2592813F5@sinclaire.sibble.net> References: <20091229052106.2592813F5@sinclaire.sibble.net> Message-ID: <291960A5CE644885B77540F79638F0AC@cune.pri> Harondel J. Sibble scribbled: > I am following the wiki page here > > http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation ... > the compilation fails as below, Zaptel is old. DAHDI is now the way to go. If you have the headers installed for the kernel you are currently using, try this. 1. http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz 2. tar -xzf dahdi-linux-complete-current.tar.gz 3. cd dahdi-linux-complete- 4. make all 5. make install 6. make config (First time only! Modify config files in /etc/dahdi) If this works, please add a DAHDI section to the wiki. If you want to use a Debian package instead of compiling the latest from scratch, try the dahdi-linux package. -- Russell Mosemann From freeswitch-list at puzzled.xs4all.nl Tue Dec 29 06:09:31 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Tue, 29 Dec 2009 15:09:31 +0100 Subject: [Freeswitch-users] problems getting openzap compiled for use withfreeswitch In-Reply-To: <291960A5CE644885B77540F79638F0AC@cune.pri> References: <20091229052106.2592813F5@sinclaire.sibble.net> <291960A5CE644885B77540F79638F0AC@cune.pri> Message-ID: <4B3A0D9B.7030708@puzzled.xs4all.nl> On 12/29/2009 01:14 PM, Russell Mosemann wrote: > Harondel J. Sibble scribbled: >> I am following the wiki page here >> >> http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation > ... >> the compilation fails as below, > > Zaptel is old. DAHDI is now the way to go. If you have the headers installed for the kernel you are currently using, try this. > > 1. http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz > 2. tar -xzf dahdi-linux-complete-current.tar.gz > 3. cd dahdi-linux-complete- > 4. make all > 5. make install > 6. make config (First time only! Modify config files in /etc/dahdi) > > If this works, please add a DAHDI section to the wiki. If you want to use a Debian package instead of compiling the latest from scratch, try the dahdi-linux package. Or you could ask Tzafrir in #asterisk where the latest and greatest dahdi deb packages are as he is afaik the maintainer. Regards, Patrick From brian at freeswitch.org Tue Dec 29 06:35:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 08:35:12 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: Hold is working fine I just tested it... I would need to see the whole dialog to see what is wrong... I tested with Polycom, Snom and Aastra. Are you doing proxy media or anything like that? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/49e690c1/attachment.html From brian at freeswitch.org Tue Dec 29 06:51:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 08:51:36 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/5412677f/attachment-0001.html From lei.tlfly at gmail.com Tue Dec 29 06:53:41 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 22:53:41 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. I will test if trunk 16055 work when I set proxy media mode to false tomorrow. 2009/12/29 Brian West > Hold is working fine I just tested it... I would need to see the whole > dialog to see what is wrong... I tested with Polycom, Snom and Aastra. > > Are you doing proxy media or anything like that? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/7c7246a0/attachment.html From lei.tlfly at gmail.com Tue Dec 29 07:10:59 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:10:59 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> Message-ID: <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> The phone I'm using is x-lite and wxCommunicator, both are sip phone software. I have not used pastebin, Is it a bug trace tool like bugzilla? Can you tell me how to register a pastbin account? 2009/12/29 Brian West > Also can you join #freeswitch-dev, include full siptrace+debug log and put > it on pastebin. > > What phone are you using? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/59271f9c/attachment-0001.html From brian at freeswitch.org Tue Dec 29 07:14:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 09:14:22 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> Message-ID: <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> the 200ok is not from FS.. its from the end point... so its not us thats not putting the SDP into the 200ok but the device you're talking to because in proxy media they are passed as is. /b On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. > I will test if trunk 16055 work when I set proxy media mode to false tomorrow. From lei.tlfly at gmail.com Tue Dec 29 07:37:22 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:37:22 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> Message-ID: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following code in sofia.c send the 200ok response sofia.c function sofia_handle_sip_i_state .... ......... switch(ss_state) ................ case nua_callstate_received: ..................... else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && !r_sdp) { nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); goto done; } The cause is r_sdp is null, but I don't known why tl_gets don't return remote sdp tag, it's quite strange. 2009/12/29 Brian West > the 200ok is not from FS.. its from the end point... so its not us thats > not putting the SDP into the 200ok but the device you're talking to because > in proxy media they are passed as is. > > /b > > On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > > > Hi Brian, thanks for your help, I am using FS in proxy media mode. the > sip agent I'm using is x-lite and wxCommunicator. > > I will test if trunk 16055 work when I set proxy media mode to false > tomorrow. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/1d4c23f7/attachment.html From lei.tlfly at gmail.com Tue Dec 29 07:38:35 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:38:35 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <50c41b4e0912290738x73d10086kcbf40973c092f2ca@mail.gmail.com> Btw, in the same scenario, FS 1.0.4 works fine. 2009/12/29 Lei Tang > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following > code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && > !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return > remote sdp tag, it's quite strange. > > 2009/12/29 Brian West > >> the 200ok is not from FS.. its from the end point... so its not us thats >> not putting the SDP into the 200ok but the device you're talking to because >> in proxy media they are passed as is. >> >> >> /b >> >> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >> >> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >> sip agent I'm using is x-lite and wxCommunicator. >> > I will test if trunk 16055 work when I set proxy media mode to false >> tomorrow. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/1bc15a3c/attachment.html From mike at jerris.com Tue Dec 29 07:39:14 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 10:39:14 -0500 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> Message-ID: <2BE60334-889A-44B9-99DE-AD1EDEBEB816@jerris.com> There is no need for you to show us traces. The fact that you are using proxy media is enough to know that the issue is with your device. If you look at the full sip trace you will see the same. Mike On Dec 29, 2009, at 10:10 AM, Lei Tang wrote: > The phone I'm using is x-lite and wxCommunicator, both are sip > phone software. > I have not used pastebin, Is it a bug trace tool like bugzilla? Can > you tell me how to register a pastbin account? > > 2009/12/29 Brian West > Also can you join #freeswitch-dev, include full siptrace+debug log > and put it on pastebin. > > What phone are you using? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > >> Hi, I think hold function in trunk 16055 is broken, I have also >> tried some old trunks, it's ok in freeswitch 1.0.4. >> The problem is, when send reponse for re-invite request, fs didn't >> send any sdp content. >> This problem is easy to reproduce, just call to fs, and press hold >> button, >> Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are >> both included. >> >> ====sip trace for trunk 16055 >> ====re-invite request sent to fs when client hold the line >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-Id: s264bdfe05129544c7e0a2c44408cb213 >> Cseq: 12860 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 462 >> Date: Tue, 29 Dec 2009 06:53:53 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport >> >> v=0 >> o=sipX 5 6 IN IP4 0.0.0.0 >> s=call >> c=IN IP4 0.0.0.0 >> t=0 0 >> m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-ID: s264bdfe05129544c7e0a2c44408cb213 >> CSeq: 12860 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> =====bad response sent by fs, sdp content is missing. >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-ID: s264bdfe05129544c7e0a2c44408cb213 >> CSeq: 12860 INVITE >> Contact: >> User-Agent: PowerIVR >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Length: 0 >> >> >> ======sip trace for fs 1.0.4 >> =====re-invite request sent to FS when client want to hold the all >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29657 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 463 >> Date: Tue, 29 Dec 2009 03:20:14 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport >> >> v=0 >> o=sipX 5 34 IN IP4 0.0.0.0 >> s=call >> c=IN IP4 0.0.0.0 >> t=0 0 >> m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29657 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> ===repsonse sent by fs, there is correct sdp content. >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29657 INVITE >> Contact: >> User-Agent: PowerIVR >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 254 >> >> v=0 >> o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 >> s=FreeSWITCH >> c=IN IP4 10.56.0.189 >> t=0 0 >> m=audio 28606 RTP/AVP 8 96 >> a=rtpmap:8 pcma/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> a=recvonly >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> Contact: >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29657 ACK >> Date: Tue, 29 Dec 2009 03:20:15 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport >> Content-Length: 0 >> >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29658 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 473 >> Date: Tue, 29 Dec 2009 03:20:18 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport >> >> v=0 >> o=sipX 5 35 IN IP4 10.56.90.223 >> s=call >> c=IN IP4 10.56.90.223 >> t=0 0 >> m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29658 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29658 INVITE >> Contact: >> User-Agent: PowerIVR >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 242 >> >> v=0 >> o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 >> s=FreeSWITCH >> c=IN IP4 10.56.0.189 >> t=0 0 >> m=audio 28606 RTP/AVP 8 96 >> a=rtpmap:8 pcma/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> Contact: >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29658 ACK >> Date: Tue, 29 Dec 2009 03:20:19 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport >> Content-Length: 0 >> >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/d25b0838/attachment-0001.html From brian at freeswitch.org Tue Dec 29 07:42:57 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 09:42:57 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <428218D4-D5C8-4AF3-B8F8-72182935CC41@freeswitch.org> Its null because the device on the other side didn't send one. We pass it as is... fix the broken device or don't use proxy media. /b On Dec 29, 2009, at 9:37 AM, Lei Tang wrote: > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return remote sdp tag, it's quite strange. From mike at jerris.com Tue Dec 29 08:08:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 11:08:08 -0500 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> This means there was no sdp sent. Did you confirm this with siptrace? On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, > following code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, > TFLAG_SDP) && !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, > TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't > return remote sdp tag, it's quite strange. > > 2009/12/29 Brian West > the 200ok is not from FS.. its from the end point... so its not us > thats not putting the SDP into the 200ok but the device you're > talking to because in proxy media they are passed as is. > > /b > > On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > > > Hi Brian, thanks for your help, I am using FS in proxy media mode. > the sip agent I'm using is x-lite and wxCommunicator. > > I will test if trunk 16055 work when I set proxy media mode to > false tomorrow. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/a6d170bc/attachment.html From anthony.minessale at gmail.com Tue Dec 29 08:45:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Dec 2009 10:45:18 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> Message-ID: <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> We now disable sofia SOA mode during proxy calls. This means that sofia will not try to get involved in the media negotiation at all which is the optimal behavior. Previous versions would butt in and try to fix the error but now it just stays out of the way. You can see in your trace that the device sends a packet with no SDP therefore so does sofia. You can either turn off proxy-media or post a bounty for me to go hack a workaround into the patch I spent many hours on getting things to work right. Whatever you experienced with 1.0.4 was a happy coincidence where sofia was fixing a bug in your phone for you. On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: > This means there was no sdp sent. Did you confirm this with siptrace? > > On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: > > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following > code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && > !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return > remote sdp tag, it's quite strange. > > 2009/12/29 Brian West < brian at freeswitch.org> > >> the 200ok is not from FS.. its from the end point... so its not us thats >> not putting the SDP into the 200ok but the device you're talking to because >> in proxy media they are passed as is. >> >> /b >> >> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >> >> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >> sip agent I'm using is x-lite and wxCommunicator. >> > I will test if trunk 16055 work when I set proxy media mode to false >> tomorrow. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/48a309e1/attachment.html From ivan at myrvold.org Tue Dec 29 09:06:45 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 18:06:45 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. Ivan Den 27. des. 2009 kl. 19.52 skrev Martin Joseph: > On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > >> Ken, >> >> The process is the same for all UNIX like platforms. You run >> >> - bootstrap.sh >> - configure >> - make >> - make install >> >> Jonathan > > > > > Actually, with the release tarballs you don't do bootstrap.sh (unless > I am mistaken). > > I have been building FreeSWITCH on OSX for quite a while (over a > year), with good results. I have NOT had any luck building from the > SVN, as it seems to throw weird errors on my problems on my platform > of choice (PPC OSX Tiger), but the released tarballs seem to work ok > (even the pre-release tarballs). > > I also think that making an OSX package of freeswitch sounds nice, > but is a bad idea, UNLESS it's set up in an automated fashion that can > stay up to date with changes. Otherwise, lazy OSX people get stuck > installing an artifact rather then the best available FreeSWITCH. This > happened with Asterisk with the Sunrise telecom people. They ended up > creating more problems then good as even years after the fact, silly > mac people where still installing the OLD compromised, buggy version > just because it was in an OSX installer... > > Hope this Helps, > Marty > > On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > >> Ken, >> >> The process is the same for all UNIX like platforms. You run >> >> - bootstrap.sh >> - configure >> - make >> - make install >> >> Jonathan >> >> On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: >> Yes, I want to set up FreeSwitch on OSX and at least see how it >> runs, assuming I can get that far. >> >> I've downloaded the latest tarball and run configure which seemed to >> complete ok. But what next? Do I actually need to run >> >> make all install sounds-install moh-install >> >> as in some lists of instructions it appears that I just need to run >> >> make >> make install >> >> Needless to say I'm not an expert at compiling although I have done >> a fair bit over the years, just not enough for it to be second >> nature. So the above apparent ambiguity puzzles me. >> >> Also, how can I compile on one machine and then actually run it on a >> different machine? Is there a relatively simple way to achieve this >> or must I manually copy all the files to the other machine. What >> files would that be? Are they all conveniently located in a single >> folder? >> >> Hopeful of some helpful advice, but let's face it, anyone doing this >> sort of thing on Christmas Eve really ought to get out more:-) >> >> >> >> Ken G i l l e t t >> >> _/_/_/_/_/_/_/_/ >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 09:11:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 11:11:40 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Ivan, I have been trying to gather up everyone to start a FreeSWITCH based softphone project for Mac, Linux and Windows... you think we could collaborate with you to accomplish this? I think if we do this right we can have a really nice phone with lots of options. Thanks, /b On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. > I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. > > Ivan > From mrene_lists at avgs.ca Tue Dec 29 10:14:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 29 Dec 2009 19:14:37 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <8DA219E3-A980-418E-A147-6AA26788E0BB@avgs.ca> This could be easily done with the Qt framework and would work nicely on osx, linux and windows. Contact me off list or on IRC (Math) if you want some help, I'd be happy to participate in such a project. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 29-Dec-09, at 6:11 PM, Brian West wrote: > Ivan, > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we > could collaborate with you to accomplish this? I think if we do > this right we can have a really nice phone with lots of options. > > Thanks, > /b > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > >> FreeSWITCH is running nicely on OS X. I have used it since July >> 2006 on my intel Macs with great success. >> I am also developing a GUI application using Cocoa. I started that >> a year ago, but haven't looked at it for a while, but this >> Christmas I have started working on it again. >> >> Ivan >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ivan at myrvold.org Tue Dec 29 11:40:52 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 20:40:52 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> Yes, I would like to participate on this. I have lot of experience with Cocoa on Mac, so I could help with that platform. But I am mostly a GUI programmer, Objective-C my language. But if this is OK with you, I would love to help out here. Ivan Den 29. des. 2009 kl. 18.11 skrev Brian West: > Ivan, > I have been trying to gather up everyone to start a FreeSWITCH based softphone project for Mac, Linux and Windows... you think we could collaborate with you to accomplish this? I think if we do this right we can have a really nice phone with lots of options. > > Thanks, > /b > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > >> FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. >> I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. >> >> Ivan >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lon at kickasspixels.com Tue Dec 29 12:01:26 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 29 Dec 2009 12:01:26 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> Message-ID: <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> I have done a little research into this for my employer. You may want to look at: http://www.qutecom.org/ - I think its QT based. http://code.google.com/p/telephone/ - Its pure Cocoa. I use this for all my testing, it lets me initiate up to 8 calls at a time. On Tue, Dec 29, 2009 at 11:40 AM, Ivan C Myrvold wrote: > Yes, I would like to participate on this. I have lot of experience with > Cocoa on Mac, so I could help with that platform. But I am mostly a GUI > programmer, Objective-C my language. > But if this is OK with you, I would love to help out here. > > Ivan > > Den 29. des. 2009 kl. 18.11 skrev Brian West: > > > Ivan, > > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we could > collaborate with you to accomplish this? I think if we do this right we can > have a really nice phone with lots of options. > > > > Thanks, > > /b > > > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > > > >> FreeSWITCH is running nicely on OS X. I have used it since July 2006 on > my intel Macs with great success. > >> I am also developing a GUI application using Cocoa. I started that a > year ago, but haven't looked at it for a while, but this Christmas I have > started working on it again. > >> > >> Ivan > >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/fc0e78a9/attachment.html From edpimentl at gmail.com Tue Dec 29 12:08:40 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 15:08:40 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> Add me to the app list .... I use MAC mostly ... Also can you list the new (better, gentler) list of commands to install FreeSwitch on a MAC OSX ... ? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/f0057d1d/attachment.html From edpimentl at gmail.com Tue Dec 29 12:18:38 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 15:18:38 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> Message-ID: <9dc4a1670912291218r581d6231wb020158e0dc1f7c1@mail.gmail.com> Here is my personal "refactored" QuteCom ... *File:* AgileSIP-MAC.zip You have 24 hours to retrieve this file at http://datrshare.datr.ws/download/e567e741e54116f488dc7a24a478e498/edpimentl%40gmail.com/AgileSIP-MAC.zip If you like it... i will create one with a FreeSwitch watermark logo for FS... Best, -E Gpro.ws DatR.ws ---------- Forwarded message ---------- From: Lon Baker Date: Tue, Dec 29, 2009 at 3:01 PM Subject: Re: [Freeswitch-users] MacOSX To: freeswitch-users at lists.freeswitch.org I have done a little research into this for my employer. You may want to look at: http://www.qutecom.org/ - I think its QT based. http://code.google.com/p/telephone/ - Its pure Cocoa. I use this for all my testing, it lets me initiate up to 8 calls at a time. On Tue, Dec 29, 2009 at 11:40 AM, Ivan C Myrvold wrote: > Yes, I would like to participate on this. I have lot of experience with > Cocoa on Mac, so I could help with that platform. But I am mostly a GUI > programmer, Objective-C my language. > But if this is OK with you, I would love to help out here. > > Ivan > > Den 29. des. 2009 kl. 18.11 skrev Brian West: > > > Ivan, > > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we could > collaborate with you to accomplish this? I think if we do this right we can > have a really nice phone with lots of options. > > > > Thanks, > > /b > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/c2a29435/attachment.html From brian at freeswitch.org Tue Dec 29 12:34:45 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 14:34:45 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> Message-ID: <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> I would love to have a FreeSWITCH based softphone for all three platforms... I just feel a project like that would be kick ass. Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like most softphones do. /b On Dec 29, 2009, at 2:08 PM, EdPimentl wrote: > Add me to the app list .... I use MAC mostly ... > > Also can you list the new (better, gentler) list of commands to install FreeSwitch on a MAC OSX ... ? > > Thanks in advance, > -E From jerry.richards at teotech.com Tue Dec 29 12:42:20 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 Dec 2009 12:42:20 -0800 Subject: [Freeswitch-users] Bypass Media True Disables MOH Message-ID: When I uncomment the following tag, internally held calls no longer hear MOH. Is there a way to have the above uncommented and still provide MOH to held calls? Best Regards, Jerry From astmac at stillnewt.org Tue Dec 29 13:08:44 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Tue, 29 Dec 2009 13:08:44 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> Message-ID: <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Sounds cool to me also. There is one softphone for OSX that doesn't suck. It's called JackenIAX. A "universal" softphone sounds cool, but not if it's dumbed down with regards to it's OSX integration. Thanks to all, Marty On Dec 29, 2009, at 12:34 PM, Brian West wrote: > I would love to have a FreeSWITCH based softphone for all three > platforms... I just feel a project like that would be kick ass. > > Must work on 32bit and 64bit of Windows, Mac and Linux ... and not > suck like most softphones do. > > /b > > On Dec 29, 2009, at 2:08 PM, EdPimentl wrote: > >> Add me to the app list .... I use MAC mostly ... >> >> Also can you list the new (better, gentler) list of commands to >> install FreeSwitch on a MAC OSX ... ? >> >> Thanks in advance, >> -E > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Dec 29 13:12:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:12:08 -0600 Subject: [Freeswitch-users] Bypass Media True Disables MOH In-Reply-To: References: Message-ID: <9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org> But it doesn't go back to bypass after.... Maybe you can post a bounty for that functionality. /b On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote: > > When I uncomment the following tag, internally held calls no longer hear > MOH. > > > > Is there a way to have the above uncommented and still provide MOH to held > calls? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Dec 29 13:14:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:14:17 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Message-ID: Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. /b On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > There is one softphone for OSX that doesn't suck. It's called > JackenIAX. From ron.freeswitch at mcleodnet.com Tue Dec 29 13:29:58 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Tue, 29 Dec 2009 13:29:58 -0800 Subject: [Freeswitch-users] Custom Events Message-ID: I have two inbound event socket sessions. I send a custom event like this on one socket: sendevent CUSTOM Event-Subclass: wpbx::Bcsm Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER and I receive the following reply: Content-Type: command/reply Reply-Text: +OK On the other event socket, I see this: Content-Length: 615 Content-Type: text/event-plain Event-Subclass: wpbx%3A%3ABcsm Event-Name: COMMAND Core-UUID: 2759f3f4-f4b7-11de-9ff0-11851d44d59f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-29%2013%3A25%3A49 Event-Date-GMT: Tue,%2029%20Dec%202009%2021%3A25%3A49%20GMT Event-Date-Timestamp: 1262121949103034 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1087 Command: sendevent%20CUSTOM Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER I was expecting to see a CUSTOM event, not COMMAND (or maybe a CUSTOM event in addition to a COMMAND event), like what I see with other custom events such as: Content-Length: 911 Content-Type: text/event-plain Event-Subclass: sofia%3A%3Aregister Event-Name: CUSTOM Core-UUID: 5d56384a-ed29-11de-85c6-27ab474dd533 FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-29%2010%3A35%3A36 Event-Date-GMT: Tue,%2029%20Dec%202009%2018%3A35%3A36%20GMT Event-Date-Timestamp: 1262111736464194 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_handle_register Event-Calling-Line-Number: 1127 Event-Subclass: sofia%3A%3Aregister profile-name: internal from-user: 698 from-host: 192.168.100.132 presence-hosts: 192.168.100.132 contact: %22user%22%20%3Csip%3A698%40192.168.100.130%3A5060%3E call-id: 1909944913%40192.168.100.130 rpid: unknown statusd: Registered(UDP) expires: 60 to-user: 698 to-host: 192.168.100.132 network-ip: 192.168.100.130 network-port: 5060 username: 698 realm: 192.168.100.132 user-agent: UTSTARCOM%20F3000/Device%20ID-F3000_TEST Am I doing something wrong, or am I missing something? Thanks, Ron From ivan at myrvold.org Tue Dec 29 13:35:08 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 22:35:08 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Message-ID: <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> I am using iSoftPhone, works great with FreeSWITCH. Ivan Den 29. des. 2009 kl. 22.14 skrev Brian West: > Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. > > /b > > On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > >> There is one softphone for OSX that doesn't suck. It's called >> JackenIAX. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 13:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:45:44 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> Message-ID: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? /b On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > I am using iSoftPhone, works great with FreeSWITCH. > > Ivan > > Den 29. des. 2009 kl. 22.14 skrev Brian West: > >> Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >> >> /b >> >> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >> >>> There is one softphone for OSX that doesn't suck. It's called >>> JackenIAX. > From ivan at myrvold.org Tue Dec 29 13:59:31 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 22:59:31 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Message-ID: <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Yes, I am for this. Happy to join such a project. Ivan Den 29. des. 2009 kl. 22.45 skrev Brian West: > Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? > > /b > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > >> I am using iSoftPhone, works great with FreeSWITCH. >> >> Ivan >> >> Den 29. des. 2009 kl. 22.14 skrev Brian West: >> >>> Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >>> >>> /b >>> >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >>> >>>> There is one softphone for OSX that doesn't suck. It's called >>>> JackenIAX. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 15:06:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 17:06:14 -0600 Subject: [Freeswitch-users] Cisco 501's Message-ID: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> Anyone have access to these phones? Two of them if possible and provisioning information? Thanks, Brian From jerry.richards at teotech.com Tue Dec 29 15:17:56 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 Dec 2009 15:17:56 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail In-Reply-To: <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> Message-ID: I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Monday, December 28, 2009 3:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/0668d901/attachment-0001.html From mike at jerris.com Tue Dec 29 15:37:10 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 18:37:10 -0500 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail In-Reply-To: References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> Message-ID: <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> try these drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz Mike On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote: > I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. > > Best Regards, > Jerry > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Monday, December 28, 2009 3:31 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail > > you have to update the sangoma driver and probably FreeSWITCH for good measure. > Its a known bug in the sangoma driver that has been fixed it the latest release. > > > > On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: > Hello All, > > I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. > > I am still having the problem where a PSTN-to-Internal call via a Sangoma > A101D card stops ringing the internal phone after about 10 seconds. It > should be ringing for 30 seconds and then go to Voice Mail (as an > Internal-to-Internal call does). > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Tuesday, December 22, 2009 8:02 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail > > > I have a Freeswitch PBX server with an installed Sangoma A101D card > connected to a PRI. Most everything works okay, however when I get an > inbound call from the PSTN, if the call is not answered within about 12 > seconds, the call ends (so it doesn't go to voice mail). If I make a call > from one internal phone to another, then it will go to voice mail after 30 > seconds. How can I get the external call to route to voice mail after 30 > seconds? > > I put a new 11595 log into the pastebin. Do you know any Freeswitch setting > that might cause this? > > If this issue has been addressed before, what string should I use to search > for it, because I can't find it. > > Thanks, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/3ac0c607/attachment.html From jmesquita at freeswitch.org Tue Dec 29 16:44:43 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 29 Dec 2009 22:44:43 -0200 Subject: [Freeswitch-users] MacOSX In-Reply-To: <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: Why don't we evolve FSGui to be a softphone? I could use a couple of experienced programmers to help out with it since I pretty much suck at it... FSGui is extensible using plugins so I think that a softphone would be nothing more then just another plugin. Math? Would you like to join in there and put a bit of work with me? Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 On Tue, Dec 29, 2009 at 7:59 PM, Ivan C Myrvold wrote: > Yes, I am for this. Happy to join such a project. > > Ivan > > Den 29. des. 2009 kl. 22.45 skrev Brian West: > > > Guessing the biggest issue is I want to create a softphone project using > FreeSWITCH as the core of the project... is this something people would be > interested in joining? > > > > /b > > > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > > > >> I am using iSoftPhone, works great with FreeSWITCH. > >> > >> Ivan > >> > >> Den 29. des. 2009 kl. 22.14 skrev Brian West: > >> > >>> Does it only do IAX? If so we'll need someone to re-write an IAX2 > stack since the libiax2 from Digium is no longer updated to keep pace with > Asterisk and is now incompatible. Which is the main reason we are thinking > about dropping IAX support unless someone writes a license compatible lib or > updates and takes over mod_iax aka owns it as their own. > >>> > >>> /b > >>> > >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > >>> > >>>> There is one softphone for OSX that doesn't suck. It's called > >>>> JackenIAX. > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/8b42de37/attachment.html From brian at freeswitch.org Tue Dec 29 16:51:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 18:51:44 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: Sounds like a plan to me... who wants to take the lead on the project... we'll host it.. setup SVN, provide jira access, fisheye and wiki space... /b On Dec 29, 2009, at 6:44 PM, Jo?o Mesquita wrote: > Why don't we evolve FSGui to be a softphone? I could use a couple of experienced programmers to help out with it since I pretty much suck at it... FSGui is extensible using plugins so I think that a softphone would be nothing more then just another plugin. > > Math? Would you like to join in there and put a bit of work with me? > > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 From edpimentl at gmail.com Tue Dec 29 17:29:51 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 20:29:51 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> For starters, one can use use the attached pics as the springboard for the future design of the softphone. -E -------------- next part -------------- An HTML attachment was scrubbed... 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Name: Picture 3.png Type: image/png Size: 32037 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/2e837688/attachment-0003.png From dujinfang at gmail.com Tue Dec 29 17:36:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 30 Dec 2009 09:36:20 +0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Message-ID: <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> I had wrote a Air based GUI, is it make sense? http://wiki.freeswitch.org/wiki/FsAir 2009/12/30 Brian West : > Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? > > /b > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > >> I am using iSoftPhone, works great with FreeSWITCH. >> >> Ivan >> >> Den 29. des. 2009 kl. 22.14 skrev Brian West: >> >>> Does it only do IAX? ?If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. ?Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >>> >>> /b >>> >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >>> >>>> There is one softphone for OSX that doesn't suck. ?It's called >>>> JackenIAX. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 17:50:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 19:50:07 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> Message-ID: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> You should join IRC and join in JM and really start the official softphone project. /b On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > I had wrote a Air based GUI, is it make sense? > > http://wiki.freeswitch.org/wiki/FsAir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/39c4ce35/attachment.html From lon at kickasspixels.com Tue Dec 29 18:30:28 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 29 Dec 2009 18:30:28 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> Message-ID: I would be happy to provide project managent, user testing and ui design to this effort. Lon Sent from my iPhone On Dec 29, 2009, at 5:50 PM, Brian West wrote: > You should join IRC and join in JM and really start the official > softphone project. > > /b > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > >> I had wrote a Air based GUI, is it make sense? >> >> http://wiki.freeswitch.org/wiki/FsAir > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/e0d04a6f/attachment.html From ron.freeswitch at mcleodnet.com Tue Dec 29 21:23:33 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Tue, 29 Dec 2009 21:23:33 -0800 Subject: [Freeswitch-users] PHP ESL Problem Message-ID: <285BD733E19541989B31B95871BF5642@fromage> Would someone please take a look at this simple PHP event socket script and tell me what I am doing wrong - or tell me that this could be a bug elsewhere? Any help would be appreciated. When I run the script without the call to execute(), everything seems fine. When I include the call to execute(), the calls to getType() return CUSTOM for a while, then later start to return the correct name. #!/usr/bin/php events('plain', 'ALL'); // call endpoint, get uuid $event = $eventSocket->api('originate', $endPoint . ' &park'); $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($dummy, $uuid) = explode('+OK ', $eventLine); if ($uuid) { break; } } // play announcement to endpoint $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav', $uuid); // monitor events while (TRUE) { echo "getType: " . $event->getType() . "\n"; $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($header, $value) = explode(': ', $eventLine); if ($header == "Event-Name") { printf($eventLine . "\n"); } if ($header == "Content-Type") { printf($eventLine . "\n"); } } printf("\n"); $event = $eventSocket->recvEvent(); }?> Run without the call to execute(): ================================== getType: CUSTOM Content-Type: api/response getType: CHANNEL_CREATE Event-Name: CHANNEL_CREATE getType: CHANNEL_OUTGOING Event-Name: CHANNEL_OUTGOING getType: CHANNEL_ORIGINATE Event-Name: CHANNEL_ORIGINATE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CHANNEL_PROGRESS Event-Name: CHANNEL_PROGRESS getType: HEARTBEAT Event-Name: HEARTBEAT getType: HEARTBEAT Event-Name: RE_SCHEDULE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CODEC Event-Name: CODEC getType: CODEC Event-Name: CODEC getType: CHANNEL_ANSWER Event-Name: CHANNEL_ANSWER getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: API Event-Name: API getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_PARK Event-Name: CHANNEL_PARK getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Run with the call to execute(): =============================== getType: CUSTOM Content-Type: command/reply getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_CREATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_OUTGOING getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ORIGINATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PROGRESS getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ANSWER getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: API getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_EXECUTE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PARK getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: COMMAND Event-Name: COMMAND getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Thanks, Ron From sharad at coraltele.com Tue Dec 29 22:17:21 2009 From: sharad at coraltele.com (Sharad) Date: Tue, 29 Dec 2009 22:17:21 -0800 (PST) Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions Message-ID: <1262153841443-4231122.post@n2.nabble.com> Hi I just want to know what should be the approx configuration of the server for 50 concurrent call sessions having 3000-4000 users. Regards -- View this message in context: http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231122.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 29 22:36:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 00:36:04 -0600 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262153841443-4231122.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> Message-ID: <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> Amplify Query... not enough data to make a logical compilation of requested data. /b On Dec 30, 2009, at 12:17 AM, Sharad wrote: > Hi > > I just want to know what should be the approx configuration of the server > for 50 concurrent call sessions having 3000-4000 users. > > Regards From sharad at coraltele.com Tue Dec 29 22:55:19 2009 From: sharad at coraltele.com (Sharad) Date: Tue, 29 Dec 2009 22:55:19 -0800 (PST) Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> Message-ID: <1262156119067-4231210.post@n2.nabble.com> Thanks /b for your kind reply... I want to integrate the freeswitch only for Auto Attendant & voicemail with a SIP server. All the users (approx 5000) shall be registered with SIP server. Whenever auto attendant or voicemail will be required, SIP server will establish the SIP trunking with Freeswitch. So I am assuming that freeswitch may get 50-60 calls at a time either for auto attendant or voicemail. Whenever there is a call with freeswitch for auto attendant, caller will punch the desired user no. & than freeswitch will throw the call back to SIP server. in this case, RTP will not be flowing through freeswitch server. So just want to know what should be the PC hardware specifications for this call traffic. I am willing to buy a good server hardware for this. So plz advice... Regards Sharad Brian West wrote: > > Amplify Query... not enough data to make a logical compilation of > requested data. > > /b > > On Dec 30, 2009, at 12:17 AM, Sharad wrote: >> Hi >> >> I just want to know what should be the approx configuration of the server >> for 50 concurrent call sessions having 3000-4000 users. >> >> Regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231210.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 29 23:19:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 01:19:53 -0600 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262156119067-4231210.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> Message-ID: <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> Might I recommend a project that was written for that exact purpose? Have you heard of OpenSIPS/OpenSER or what is that other name... I can't spell it nor remember it... but it combined with FreeSWITCH would solve your problem better I suspect. Use the proxy for registrations and FreeSWITCH for media servers. /b On Dec 30, 2009, at 12:55 AM, Sharad wrote: > > Thanks /b for your kind reply... > > I want to integrate the freeswitch only for Auto Attendant & voicemail with > a SIP server. All the users (approx 5000) shall be registered with SIP > server. Whenever auto attendant or voicemail will be required, SIP server > will establish the SIP trunking with Freeswitch. > > So I am assuming that freeswitch may get 50-60 calls at a time either for > auto attendant or voicemail. Whenever there is a call with freeswitch for > auto attendant, caller will punch the desired user no. & than freeswitch > will throw the call back to SIP server. in this case, RTP will not be > flowing through freeswitch server. > > So just want to know what should be the PC hardware specifications for this > call traffic. > > I am willing to buy a good server hardware for this. So plz advice... > > Regards > Sharad From talk2ram at gmail.com Tue Dec 29 23:20:38 2009 From: talk2ram at gmail.com (ram) Date: Wed, 30 Dec 2009 12:50:38 +0530 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262156119067-4231210.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> Message-ID: how about this URL http://wiki.freeswitch.org/wiki/Specsheet#Minimum.2FRecommended_System_Requirements Ram On Wed, Dec 30, 2009 at 12:25 PM, Sharad wrote: > > Thanks /b for your kind reply... > > I want to integrate the freeswitch only for Auto Attendant & voicemail with > a SIP server. All the users (approx 5000) shall be registered with SIP > server. Whenever auto attendant or voicemail will be required, SIP server > will establish the SIP trunking with Freeswitch. > > So I am assuming that freeswitch may get 50-60 calls at a time either for > auto attendant or voicemail. Whenever there is a call with freeswitch for > auto attendant, caller will punch the desired user no. & than freeswitch > will throw the call back to SIP server. in this case, RTP will not be > flowing through freeswitch server. > > So just want to know what should be the PC hardware specifications for this > call traffic. > > I am willing to buy a good server hardware for this. So plz advice... > > Regards > Sharad > > > > Brian West wrote: > > > > Amplify Query... not enough data to make a logical compilation of > > requested data. > > > > /b > > > > On Dec 30, 2009, at 12:17 AM, Sharad wrote: > >> Hi > >> > >> I just want to know what should be the approx configuration of the > server > >> for 50 concurrent call sessions having 3000-4000 users. > >> > >> Regards > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231210.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/b3318f36/attachment.html From talk2ram at gmail.com Wed Dec 30 00:19:57 2009 From: talk2ram at gmail.com (ram) Date: Wed, 30 Dec 2009 13:49:57 +0530 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> Message-ID: On Wed, Dec 30, 2009 at 12:49 PM, Brian West wrote: > Might I recommend a project that was written for that exact purpose? Have > you heard of OpenSIPS/OpenSER or what is that other name... I can't spell it > nor remember it... but it combined with FreeSWITCH would solve your problem > better I suspect. Use the proxy for registrations and FreeSWITCH for media > servers. > > /b > yes you can use Opensips to Loadbalance N (X) Freeswitch Boxes for loadbalance to meet high volume calls Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a4abc2a6/attachment.html From msc at freeswitch.org Wed Dec 30 00:43:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 00:43:16 -0800 Subject: [Freeswitch-users] Cisco 501's In-Reply-To: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> References: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> Message-ID: <87f2f3b90912300043t111582b9y602aeb88f4ea9126@mail.gmail.com> I have one in my garage collecting dust. :) On Tue, Dec 29, 2009 at 3:06 PM, Brian West wrote: > Anyone have access to these phones? Two of them if possible and > provisioning information? > > Thanks, > Brian > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/7c8d0b8e/attachment.html From mcampbellsmith at gmail.com Wed Dec 30 01:42:09 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 30 Dec 2009 20:42:09 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> Message-ID: <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> Hi Brian, But that is all I see... I don't see any further messages after the 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that this is part of the authentication procedure, but it stops at the 401/Authorized message What about in the pastebin? Do you see anything there? This is when the call gets setup, but the IP address in the SDP says a private IP address http://pastebin.freeswitch.org/11632 On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: > If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. ?I would have to see what happens after the 401 to see if it really did fail. > > /b > > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > >> This is all I see and then registration fails. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From codecomplete at free.fr Wed Dec 30 03:23:17 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 03:23:17 -0800 (PST) Subject: [Freeswitch-users] Port item in "sofia status" output? Message-ID: <26965939.post@talk.nabble.com> Hello I was wondering: What is the meaning of the Port item that shows up when I run "sofia status"? > sofia status profile internal reg [...] IP: 192.168.0.1 Port: 59724 Is is the UDP port on which the phone listens for incoming SIP messages, or is it the RTP port on which it expects to receive voice packets from the remote phone to which is it will be connected? BTW, does RTP use a single port to TX/RX voice, or does it use the RTP port and the one that immediately follows? Thank you. -- View this message in context: http://old.nabble.com/Port-item-in-%22sofia-status%22-output--tp26965939p26965939.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From alhakeem at gmail.com Wed Dec 30 03:38:10 2009 From: alhakeem at gmail.com (Abdul Hakeem) Date: Wed, 30 Dec 2009 11:38:10 -0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> Message-ID: <001801ca8944$91036260$b30a2720$@com> Is anyone working on this or similar soft-phone project ? Please share with us ? Cheers, AH From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of EdPimentl Sent: Wednesday, December 30, 2009 1:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] MacOSX For starters, one can use use the attached pics as the springboard for the future design of the softphone. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/d1ccc908/attachment.html From codecomplete at free.fr Wed Dec 30 04:03:06 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:03:06 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> References: <26892767.post@talk.nabble.com> <26903707.post@talk.nabble.com> <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> Message-ID: <26966693.post@talk.nabble.com> Thanks for the tip. I ended up re-installing CentOS followed by compiling the latest SVN source, and I'm back to normal. The only possible cause I see is that changing the PSU sometimes requires flushing the CMOS, but as to why this would cause Freeswitch to route calls to VM, I have no idea. Sorry for the confusion. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26966693.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 30 04:07:38 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:07:38 -0800 (PST) Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <26966731.post@talk.nabble.com> I'll take advantage of this thread to ask whether it's OK/recommended to create a new user/group "freeswitch", and "chown -Rf /usr/local/freeswitch"? Thank you. -- View this message in context: http://old.nabble.com/freeswitch-init-tp26926152p26966731.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 30 04:18:35 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:18:35 -0800 (PST) Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch In-Reply-To: <20091229052106.2592813F5@sinclaire.sibble.net> References: <20091229052106.2592813F5@sinclaire.sibble.net> Message-ID: <26966821.post@talk.nabble.com> Harondel J. Sibble wrote: > I am trying to get 2x X100P's working Provided you are aware that a lot of users have problems getting X100xP cards to work on their hardware (either due to some issue with the PCI bus, or the Silicon Labs DAA chips not supporting the POTS line to which they are connected)... here's what I did to compile Dahdi (the new name for Zaptel) from source: 1. Download and unpack the Dahdi tarball 2. make all ; make install ; make config 3. cd /etc/dahdi/ ; vi system.conf: #Per FS wiki: "Tones should be configured in OpenZAP and not in zaptel. FreeSWITCH uses its libteletone for tones generation and detection, and does not rely on zaptel tones configuration. Therefore it does not matter which country zone is configured in zaptel. Make sure that you have loaded your country-specific tones at /etc/openzap/tones.conf" #loadzone = fr #defaultzone = fr fxsks=1 4. vi modules; add "wctdm" 5. /etc/init.d/dahdi start 6. lsmod to check loaded modules 7. dahdi_cfg -vvv ; dahdi_scan ; dahdi_test -v 8. ls -la /proc/dahdi/ ; cat /proc/dahdi/1 The next step, which I haven't done yet, is compiling OpenZap so it talks to Dahdi/Zaptel, which talks to the actual hardware. -- View this message in context: http://old.nabble.com/problems-getting-openzap-compiled-for-use-with-freeswitch-tp26951344p26966821.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From petedao at gmail.com Wed Dec 30 05:14:03 2009 From: petedao at gmail.com (Pete Kay) Date: Wed, 30 Dec 2009 21:14:03 +0800 Subject: [Freeswitch-users] freeswitch and H323 Message-ID: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> Hi, has anyone been able to get H323 to work? I have problem trying to get it compiled with either 1.0.4 or 1.0.5. Thanks, pete From Russell.Mosemann at cune.org Wed Dec 30 05:57:43 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Wed, 30 Dec 2009 07:57:43 -0600 Subject: [Freeswitch-users] freeswitch init In-Reply-To: <26966731.post@talk.nabble.com> References: <26966731.post@talk.nabble.com> Message-ID: <953218C0A63749C4BF083829F5372E5E@cune.pri> Fred-145 wrote: > I'll take advantage of this thread to ask whether it's OK/recommended to > create a new user/group "freeswitch", and "chown -Rf > /usr/local/freeswitch"? Yes. -- Russell Mosemann From anthony.minessale at gmail.com Wed Dec 30 06:04:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 08:04:39 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> Message-ID: <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> are you using more than one profile here? if so can you repeat the trace with siptrace on in both profiles. I notice this on the trace: 1. Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 2. From: 1000 ;tag=70c4b62bd9443c9fo0 and this: sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000 at 192.168.1.120] 192.168.1 .120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20 Are you behind double nat and or is FS also behind nat? To address the gentleman who mentioned he was spoiled by asterisk. Keep in mind we have several advanced nat techniques but many of them break other situations and they are not all turned on at once with one parameter like in asterisk, we prefer you use the appropriate ones based on the situation. On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi Brian, > > But that is all I see... I don't see any further messages after the > 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that > this is part of the authentication procedure, but it stops at the > 401/Authorized message > > What about in the pastebin? Do you see anything there? This is when > the call gets setup, but the IP address in the SDP says a private IP > address http://pastebin.freeswitch.org/11632 > > On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: > > If you're using the 401 as an indication that it fails then you don't > understand how digest authentication works. I would have to see what > happens after the 401 to see if it really did fail. > > > > /b > > > > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > > > >> This is all I see and then registration fails. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/3cd5a327/attachment.html From dujinfang at gmail.com Wed Dec 30 06:17:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 30 Dec 2009 22:17:39 +0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> Message-ID: <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> I rarely joined in IRC, becuase I live in China, timezone +8000 .... I really would like to start the official softphone, btw, what is JM? 2009/12/30, Brian West : > You should join IRC and join in JM and really start the official softphone > project. > > /b > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > >> I had wrote a Air based GUI, is it make sense? >> >> http://wiki.freeswitch.org/wiki/FsAir > > From jmesquita at freeswitch.org Wed Dec 30 06:27:40 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 30 Dec 2009 12:27:40 -0200 Subject: [Freeswitch-users] MacOSX In-Reply-To: <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> Message-ID: What is JM is not the question but rather WHO is JM and that would be me. :-) I have already stripped down the config handler based on mod_xml_curl. I have been discussing with Brian how to make it happen and I am conducting a couple of tests with Qt. Today I might be able to have it properly linked with Qt and the core spawn on its own thread inside the Qt event loop. I'll keep you posted. Jo?o Mesquita A.K.A -> JM On Wed, Dec 30, 2009 at 12:17 PM, Seven Du wrote: > I rarely joined in IRC, becuase I live in China, timezone +8000 .... > I really would like to start the official softphone, btw, what is JM? > > 2009/12/30, Brian West : > > You should join IRC and join in JM and really start the official > softphone > > project. > > > > /b > > > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > > > >> I had wrote a Air based GUI, is it make sense? > >> > >> http://wiki.freeswitch.org/wiki/FsAir > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cad80cc2/attachment.html From lei.tlfly at gmail.com Wed Dec 30 06:38:44 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 30 Dec 2009 22:38:44 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> Message-ID: <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> Hi Anthony and Brian, I have grep some sip trace log, Could you please take a look at the attachments? lega.pcap is the log of a leg, legb.pcap is the log of b leg. It seems fs never sent the reinvite request to b, and sent 200ok response without sdp content to a immediately. My scenario is as follow, and I'm using proxy media mode. invite invite a <------------> FS<---------------> b 2009/12/30 Anthony Minessale > We now disable sofia SOA mode during proxy calls. > This means that sofia will not try to get involved in the media negotiation > at all which is the optimal behavior. > Previous versions would butt in and try to fix the error but now it just > stays out of the way. > > You can see in your trace that the device sends a packet with no SDP > therefore so does sofia. > > You can either turn off proxy-media or post a bounty for me to go hack a > workaround into the patch I spent many hours on getting things to work > right. Whatever you experienced with 1.0.4 was a happy coincidence where > sofia was fixing a bug in your phone for you. > > > > > On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: > >> This means there was no sdp sent. Did you confirm this with siptrace? >> >> On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: >> >> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following >> code in sofia.c send the 200ok response >> sofia.c >> function sofia_handle_sip_i_state .... >> ......... >> switch(ss_state) >> ................ >> case nua_callstate_received: >> ..................... >> else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && >> !r_sdp) { >> nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); >> sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); >> goto done; >> } >> >> The cause is r_sdp is null, but I don't known why tl_gets don't return >> remote sdp tag, it's quite strange. >> >> 2009/12/29 Brian West < brian at freeswitch.org> >> >>> the 200ok is not from FS.. its from the end point... so its not us thats >>> not putting the SDP into the 200ok but the device you're talking to because >>> in proxy media they are passed as is. >>> >>> /b >>> >>> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >>> >>> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >>> sip agent I'm using is x-lite and wxCommunicator. >>> > I will test if trunk 16055 work when I set proxy media mode to false >>> tomorrow. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: lega.pcap Type: application/octet-stream Size: 15372 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: legb.pcap Type: application/octet-stream Size: 9395 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0003.obj From anthony.minessale at gmail.com Wed Dec 30 07:19:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 09:19:26 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> Message-ID: <191c3a030912300719m5557ad3cyad1f483643001f67@mail.gmail.com> I was wondering if you read my last email? your phone is not sending an SDP, but you expect FS to magically pass it on with an SDP? Did you try anything I said? I feel that you have ignored my 2 paragraph explanation. On Wed, Dec 30, 2009 at 8:38 AM, Lei Tang wrote: > Hi Anthony and Brian, I have grep some sip trace log, Could you please take > a look at the attachments? lega.pcap is the log of a leg, legb.pcap is the > log of b leg. It seems fs never sent the reinvite request to b, and sent > 200ok response without sdp content to a immediately. > My scenario is as follow, and I'm using proxy media mode. > > invite invite > a <------------> FS<---------------> b > > 2009/12/30 Anthony Minessale > > We now disable sofia SOA mode during proxy calls. >> This means that sofia will not try to get involved in the media >> negotiation at all which is the optimal behavior. >> Previous versions would butt in and try to fix the error but now it just >> stays out of the way. >> >> You can see in your trace that the device sends a packet with no SDP >> therefore so does sofia. >> >> You can either turn off proxy-media or post a bounty for me to go hack a >> workaround into the patch I spent many hours on getting things to work >> right. Whatever you experienced with 1.0.4 was a happy coincidence where >> sofia was fixing a bug in your phone for you. >> >> >> >> >> On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: >> >>> This means there was no sdp sent. Did you confirm this with siptrace? >>> >>> On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: >>> >>> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, >>> following code in sofia.c send the 200ok response >>> sofia.c >>> function sofia_handle_sip_i_state .... >>> ......... >>> switch(ss_state) >>> ................ >>> case nua_callstate_received: >>> ..................... >>> else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && >>> !r_sdp) { >>> nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); >>> sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); >>> goto done; >>> } >>> >>> The cause is r_sdp is null, but I don't known why tl_gets don't return >>> remote sdp tag, it's quite strange. >>> >>> 2009/12/29 Brian West < brian at freeswitch.org> >>> >>>> the 200ok is not from FS.. its from the end point... so its not us thats >>>> not putting the SDP into the 200ok but the device you're talking to because >>>> in proxy media they are passed as is. >>>> >>>> /b >>>> >>>> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >>>> >>>> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >>>> sip agent I'm using is x-lite and wxCommunicator. >>>> > I will test if trunk 16055 work when I set proxy media mode to false >>>> tomorrow. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Lei.Tang >>> lei.tlfly at gmail.com >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/5c16463a/attachment.html From codecomplete at free.fr Wed Dec 30 07:29:12 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 07:29:12 -0800 (PST) Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B3142F7.1080600@skypro.be> References: <4B3142F7.1080600@skypro.be> Message-ID: <26968615.post@talk.nabble.com> Kristoff Bonne-2 wrote: > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) It looks like one of those ATA boxes that has an FXO port so you can make calls either through a VoIP provider or through your landline. Now, if it can also act as an SIP/PSTN gateway, I'd be very interested in that ?15 piece of hardware ;-) -- View this message in context: http://old.nabble.com/tigerjet-560C-USB-to-rj11%3A-incorperate-usbhid-usbsnd-device-into-freeswitch-tp26895402p26968615.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From fdenkens at ilibris.be Wed Dec 30 08:15:33 2009 From: fdenkens at ilibris.be (Frederik Denkens | iLibris) Date: Wed, 30 Dec 2009 17:15:33 +0100 Subject: [Freeswitch-users] Freeswitch 1.0.4 and Patton 4554 gateway registration problem Message-ID: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> Hi all! We our looking forward to replacing our legacy PBX with a Freeswitch platform, but are struggling with getting our Patton SmartNode 4554 (SIP/ISDN) gateway hooked up. Any assistance from the community would be great! First step is getting Freeswitch and the Patton talking to each other. Goal? To be able to have incoming and outgoing calls going over the Patton to the ISDN network. Internal PBX: - Freeswitch 1.0.4 installed from tar with default (demo) setup on IP 10.156.10.93 - Patton 4554 with simple config on IP 10.156.10.90 We set it up that the Patton registers with the PBX and we get the error: --------------------- 2009-12-30 17:07:08.11907 [WARNING] sofia_reg.c:1771 Can't find user [101 at 10.156.10.93 ] You must define a domain called '10.156.10.93' in your directory and add a user with the id="101" attribute and you must configure your device to use the proper domain in it's authentication credentials. --------------------- Find more info below. So any help would be great! Many thanks! Frederik Denkens Belgium +32 475 96 04 93 We defined a gateway in conf/sip_profiles/external/patton.xml: --------------------- --> --------------------- And the relevant parts of the Patton config: --------------------- # define auth authentication-service AUTH_SVC username 101 password 101 # patton registers location-service LOCATION_SVC domain 1 10.156.10.93 identity 101 authentication outbound authenticate 1 authentication-service AUTH_SVC username 101 registration outbound registrar 10.156.10.93 5080 lifetime 3600 register auto context sip-gateway GW_SIP interface IF_SIP bind interface IF_IP_WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown --------------------- Output from 'sofia status' --------------------- 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== internal profile sip:mod_sofia at 10.156.10.93:5060 RUNNING (0) external profile sip:mod_sofia at 10.156.10.93:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG 10.156.10.93 alias internal ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== --------------------- sip trace --------------------- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.156.10.90:5060;branch=z9hG4bKe231fe7b97aa2c0cf From: ;tag=b793adfadf To: ;tag=Ua16yjtFgDQ5c Call-ID: add8fbd86264310b CSeq: 12517 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 Before printing this e-mail, please consider the impact on the environment. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/7be51e9e/attachment-0001.html From voippa at gmail.com Wed Dec 30 06:19:54 2009 From: voippa at gmail.com (David Schwartz) Date: Wed, 30 Dec 2009 16:19:54 +0200 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? Message-ID: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> What is the current communication protocol with the media proxy? I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? Thanks, Adino -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/ead6076d/attachment.html From voippa at gmail.com Wed Dec 30 08:09:07 2009 From: voippa at gmail.com (Adino) Date: Wed, 30 Dec 2009 18:09:07 +0200 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? Message-ID: <6e40a4420912300809t26d1d316had55361de65d2f27@mail.gmail.com> What is the current communication protocol with the media proxy? I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/87546a48/attachment.html From brian at freeswitch.org Wed Dec 30 08:23:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 10:23:15 -0600 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? In-Reply-To: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> References: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> Message-ID: <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> Someone could pay to have support added but as it stands we don't have it yet. If its going to be done it has to be proper mgcp support not half ass support. /b On Dec 30, 2009, at 8:19 AM, David Schwartz wrote: > I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? From help at pdscc.com Wed Dec 30 08:28:48 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Wed, 30 Dec 2009 08:28:48 -0800 Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch In-Reply-To: <26966821.post@talk.nabble.com> References: <20091229052106.2592813F5@sinclaire.sibble.net>, <26966821.post@talk.nabble.com> Message-ID: <20091230162847.DAC71174C@sinclaire.sibble.net> On 30 Dec 2009 at 4:18, Fred-145 wrote: > Provided you are aware that a lot of users have problems getting > X100xP cards to work on their hardware (either due to some issue with > the PCI bus, or the Silicon Labs DAA chips not supporting the POTS line > to which they are connected)... here's what I did to compile Dahdi (the > new name for Zaptel) from source: Not an issue in this machine, the 2 cards worked fine with asterisk now and dahdi previously. > The next step, which I haven't done yet, is compiling OpenZap so it > talks to Dahdi/Zaptel, which talks to the actual hardware. I'd love to hear your feedback on that when you do. I'm hoping to get this up and running this weekend, but currently more committed to getting zrtp working with the tivi client on my windows mobile phone. It's giving a registration forbidden message, but I don't see anything on the fs_cli to indicate it's even connected :-( -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From msc at freeswitch.org Wed Dec 30 09:00:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 09:00:18 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Message-ID: <87f2f3b90912300900i678303e3ofcbba57fce2a0faa@mail.gmail.com> Please join us today for a special Wednesday edition of the weekly FreeSWITCH conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30 -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cd533d43/attachment.html From ron.freeswitch at mcleodnet.com Wed Dec 30 09:33:34 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Wed, 30 Dec 2009 09:33:34 -0800 Subject: [Freeswitch-users] Custom Events In-Reply-To: References: Message-ID: <754D535B4FC64203B1F8D1670F360F3E@fromage> I found that this works as a work-around: sendevent Event-Name: CUSTOM Event-Subclass: wpbx::Bcsm Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Tuesday, December 29, 2009 1:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Custom Events > > I have two inbound event socket sessions. > > I send a custom event like this on one socket: > > sendevent CUSTOM > Event-Subclass: wpbx::Bcsm > Bcsm-Operation: Bcsm-Event > Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d > Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 > Bcsm-Event: ANSWER > > and I receive the following reply: > > Content-Type: command/reply > Reply-Text: +OK > > > On the other event socket, I see this: > > Content-Length: 615 > Content-Type: text/event-plain > > Event-Subclass: wpbx%3A%3ABcsm > Event-Name: COMMAND > Core-UUID: 2759f3f4-f4b7-11de-9ff0-11851d44d59f > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-29%2013%3A25%3A49 > Event-Date-GMT: Tue,%2029%20Dec%202009%2021%3A25%3A49%20GMT > Event-Date-Timestamp: 1262121949103034 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: read_packet > Event-Calling-Line-Number: 1087 > Command: sendevent%20CUSTOM > Bcsm-Operation: Bcsm-Event > Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d > Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 > Bcsm-Event: ANSWER > > > I was expecting to see a CUSTOM event, not COMMAND (or maybe a CUSTOM > event > in addition to a COMMAND event), like what I see with other custom events > such as: > > Content-Length: 911 > Content-Type: text/event-plain > > Event-Subclass: sofia%3A%3Aregister > Event-Name: CUSTOM > Core-UUID: 5d56384a-ed29-11de-85c6-27ab474dd533 > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-29%2010%3A35%3A36 > Event-Date-GMT: Tue,%2029%20Dec%202009%2018%3A35%3A36%20GMT > Event-Date-Timestamp: 1262111736464194 > Event-Calling-File: sofia_reg.c > Event-Calling-Function: sofia_reg_handle_register > Event-Calling-Line-Number: 1127 > Event-Subclass: sofia%3A%3Aregister > profile-name: internal > from-user: 698 > from-host: 192.168.100.132 > presence-hosts: 192.168.100.132 > contact: %22user%22%20%3Csip%3A698%40192.168.100.130%3A5060%3E > call-id: 1909944913%40192.168.100.130 > rpid: unknown > statusd: Registered(UDP) > expires: 60 > to-user: 698 > to-host: 192.168.100.132 > network-ip: 192.168.100.130 > network-port: 5060 > username: 698 > realm: 192.168.100.132 > user-agent: UTSTARCOM%20F3000/Device%20ID-F3000_TEST > > > Am I doing something wrong, or am I missing something? > > Thanks, > Ron > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com From ken at ukgb.net Wed Dec 30 10:45:38 2009 From: ken at ukgb.net (Ken Gillett) Date: Wed, 30 Dec 2009 18:45:38 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> Message-ID: This is beginning to confuse me. Some say just: > - configure > - make > - make install is required, but the docs say more is needed for modules.conf. I'm still not sure if this only applies when modules.conf has been edited. Anyone help there? On 28 Dec 2009, at 14:37, Brian West wrote: > "all" is no longer needed. > > /b > > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > >> make all install sounds-install moh-install. So make install sounds-install moh-install. is required? Always? Why? Also, to bring this topic back to my original question (not that the diversity hasn't been interesting:-) How can I best compile FS on one Mac and install it onto a different Mac? Ken G i l l e t t _/_/_/_/_/_/_/_/ From jaugenstine at gmail.com Wed Dec 30 11:03:34 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Wed, 30 Dec 2009 11:03:34 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> Message-ID: <207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com> Ken, configure make make install This sequence of steps builds and installs the default configuration but without the audio files. If you want the sound files installed also then: make install sounds-install moh-install Now the default sound files for conferencing, voicemail and music on hold are installed. If you want to modify the default install to customize the build you can add and remove modules in modules.conf. Then you run make/make install again to build those modules that are now included in the edited modules.conf file. Jonathan On Wed, Dec 30, 2009 at 10:45 AM, Ken Gillett wrote: > This is beginning to confuse me. Some say just: > > > - configure > > - make > > - make install > > is required, but the docs say more is needed for modules.conf. I'm still > not sure if this only applies when modules.conf has been edited. Anyone help > there? > > On 28 Dec 2009, at 14:37, Brian West wrote: > > > "all" is no longer needed. > > > > /b > > > > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > > > >> make all install sounds-install moh-install. > > So > > make install sounds-install moh-install. > > is required? Always? Why? > > Also, to bring this topic back to my original question (not that the > diversity hasn't been interesting:-) > > How can I best compile FS on one Mac and install it onto a different Mac? > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/78a73fd1/attachment.html From jerry.richards at teotech.com Wed Dec 30 11:16:31 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 11:16:31 -0800 Subject: [Freeswitch-users] SVN Trunk Release Label Message-ID: When you generate a release or pre-release, how can I find out what subversion trunk revision # it corresponds to? Would it make sense to make a separate subversion revision with a comment declaring the release/pre-release version (e.g. "1.0.5pre9", or "1.0.5")? Best Regards, Jerry From brian at freeswitch.org Wed Dec 30 11:49:38 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 13:49:38 -0600 Subject: [Freeswitch-users] SVN Trunk Release Label In-Reply-To: References: Message-ID: <4621070F-27B3-4D7D-B85A-D4C47F940115@freeswitch.org> You'll see an official announcement... to find the latest pre of 1.0.5 go to http://latest.freeswitch.org/ /b On Dec 30, 2009, at 1:16 PM, Jerry Richards wrote: > When you generate a release or pre-release, how can I find out what > subversion trunk revision # it corresponds to? Would it make sense to make > a separate subversion revision with a comment declaring the > release/pre-release version (e.g. "1.0.5pre9", or "1.0.5")? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Wed Dec 30 12:13:13 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 31 Dec 2009 07:13:13 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> Message-ID: <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> Hi Anthony, The only profiles I have defined are external and internal. These should be using internal... 192.168.1.120 is the FS box, which is NAT'd. Never had any problems with this being NAT'd though 192.168.1.121 is a PAP2 ATA connected to FS I don't use proxy media. I am trying to call an SPA3102, which is on the internet and NAT'd (external IP address 11.11.11.11 in the trace and internal/private ip address of 192.168.1.3). Thanks! On Thu, Dec 31, 2009 at 1:04 AM, Anthony Minessale wrote: > are you using more than one profile here? if so can you repeat the trace > with siptrace on in both profiles. > > I notice this on the trace: > > ? ?Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 > ? ?From: 1000 ;tag=70c4b62bd9443c9fo0 > > and this: > > sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000 at 192.168.1.120] > 192.168.1.120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20 > > Are you behind double nat and or is FS also behind nat? > > To address the gentleman who mentioned he was spoiled by asterisk.? Keep in > mind we have several advanced nat techniques but many of them break other > situations and they are not all turned on at once with one parameter like in > asterisk, we prefer you use the appropriate ones based on the situation. > > > > > On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith > wrote: >> >> Hi Brian, >> >> But that is all I see... I don't see any further messages after the >> 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that >> this is part of the authentication procedure, but it stops at the >> 401/Authorized message >> >> What about in the pastebin? ?Do you see anything there? ?This is when >> the call gets setup, but the IP address in the SDP says a private IP >> address ?http://pastebin.freeswitch.org/11632 >> >> On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: >> > If you're using the 401 as an indication that it fails then you don't >> > understand how digest authentication works. ?I would have to see what >> > happens after the 401 to see if it really did fail. >> > >> > /b >> > >> > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: >> > >> >> This is all I see and then registration fails. >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Dec 30 12:21:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 14:21:16 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> Message-ID: <7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please. /b On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote: > Hi Anthony, > > The only profiles I have defined are external and internal. These > should be using internal... > > 192.168.1.120 is the FS box, which is NAT'd. Never had any problems > with this being NAT'd though > 192.168.1.121 is a PAP2 ATA connected to FS > > I don't use proxy media. > > I am trying to call an SPA3102, which is on the internet and NAT'd > (external IP address 11.11.11.11 in the trace and internal/private ip > address of 192.168.1.3). > > Thanks! From jerry.richards at teotech.com Wed Dec 30 14:47:42 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 14:47:42 -0800 Subject: [Freeswitch-users] FS Reporting Previous Presence Status, Not Current Presence Status Message-ID: I'm using two Bria softphones and I found that if I change presence a few times in succession (separated by several seconds), FS is reporting the previous status, not the current change. For example, if I change the presence from Available to Away to Available, the last PUBLISH sent from the softphone indicates Available, but then FS sends a NOTIFY with Away to the subscriber(s). Is this a bug? Best Regards, Jerry From anthony.minessale at gmail.com Wed Dec 30 15:00:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 17:00:51 -0600 Subject: [Freeswitch-users] FS Reporting Previous Presence Status, Not Current Presence Status In-Reply-To: References: Message-ID: <191c3a030912301500h75fa4cbeh1bb5dda249983df3@mail.gmail.com> Since its a commercial phone, we don't have it so we could not debug it, so it's hard to answer your question. On Wed, Dec 30, 2009 at 4:47 PM, Jerry Richards wrote: > I'm using two Bria softphones and I found that if I change presence a few > times in succession (separated by several seconds), FS is reporting the > previous status, not the current change. > > For example, if I change the presence from Available to Away to Available, > the last PUBLISH sent from the softphone indicates Available, but then FS > sends a NOTIFY with Away to the subscriber(s). Is this a bug? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/0eb19da5/attachment.html From jerry.richards at teotech.com Wed Dec 30 15:23:49 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 15:23:49 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get RoutedtoVoice Mail In-Reply-To: <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com><191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> Message-ID: Mike, Yes your updated driver works correctly. This is very cool. Thanks! Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Tuesday, December 29, 2009 3:37 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get RoutedtoVoice Mail try these drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz Mike On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote: I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Monday, December 28, 2009 3:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cd511a1d/attachment-0001.html From vmknott at gmail.com Wed Dec 30 15:24:45 2009 From: vmknott at gmail.com (VM Knott) Date: Wed, 30 Dec 2009 18:24:45 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) Message-ID: I am attempting to use multiple FreeSWITCH servers to share a common collection of voicemail boxes. Everything is working great, except for when it comes to managing the custom greetings within the individual voicemail boxes. As I understand it, when creating a custom greeting within a voicemail box, a record is stored in the voicemail_default.db file located in the ../freeswitch/db directory (which allows FS to know where to retrieve the greeting message). My original plan was to link (NFS) the remote servers to the directory to reference this file, and keep the custom greetings in an additional shared directory location (again, all of this working great so far). voicemail_prefs cols: username, domain, name_path, greeting_path, password Unfortunately, the record that is inserted into this file includes an IP address (domain) of the FS server handling the call. This complicates things if I am ?hot swapping? FS servers in and out of the server cluster. Each time I add a server to the cluster, I will have to have a process go into this file and replicate all of the records for each mailbox account, to account for the specific server ip address. Seems awkward. Am I approaching this wrong? i.e., is there a way to by pass this voicemail_prefs table? or would I have to dig into the mod_voicemail.c source code and customize it? - VMK From msc at freeswitch.org Wed Dec 30 16:02:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 16:02:16 -0800 Subject: [Freeswitch-users] Special Announcement: Latest FreeSWITCH Files Message-ID: <87f2f3b90912301602u4697b0ccjf89e6ce89d8a6fec@mail.gmail.com> Greetings all! We would like to let everyone know that we have a new place for you to download the latest FreeSWITCH source packages: http://latest.freeswitch.org. The official announcement can be read here. Thanks for all your help in making FreeSWITCH a great project and a great community. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/6d5adb95/attachment.html From msc at freeswitch.org Wed Dec 30 16:23:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 16:23:42 -0800 Subject: [Freeswitch-users] Freeswitch 1.0.4 and Patton 4554 gateway registration problem In-Reply-To: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> References: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> Message-ID: <87f2f3b90912301623y1c0e21chf60e4f34aa78c7fa@mail.gmail.com> You definitely need to got on the latest FreeSWITCH: http://latest.freeswitch.org or use SVN. Secondly, if you are seeing this message then you've probably not got your domain configured. Is 10.156.10.93 your FreeSWITCH server? If you are using the default configs then the domain will be the IP address of your FS server. Go to the FS command line and do "eval ${domain}" and see what it says. Make sure that you don't have any strange things happening like having multiple NICs, etc. that might be wreaking havoc on your system. -MC On Wed, Dec 30, 2009 at 8:15 AM, Frederik Denkens | iLibris < fdenkens at ilibris.be> wrote: > Hi all! > > We our looking forward to replacing our legacy PBX with a Freeswitch > platform, but are struggling with getting our Patton SmartNode 4554 > (SIP/ISDN) gateway hooked up. Any assistance from the community would be > great! > > First step is getting Freeswitch and the Patton talking to each other. > > Goal? To be able to have incoming and outgoing calls going over the Patton > to the ISDN network. > > Internal PBX: > - Freeswitch 1.0.4 installed from tar with default (demo) setup on IP > 10.156.10.93 > - Patton 4554 with simple config on IP 10.156.10.90 > > We set it up that the Patton registers with the PBX and we get the error: > > --------------------- > *2009-12-30 17:07:08.11907 [WARNING] sofia_reg.c:1771 Can't find user [ > 101 at 10.156.10.93]* > *You must define a domain called '10.156.10.93' in your directory and add > a user with the id="101" attribute* > *and you must configure your device to use the proper domain in it's > authentication credentials.* > --------------------- > > Find more info below. > > So any help would be great! > > Many thanks! > > Frederik Denkens > Belgium > *+32 475 96 04 93 * > > > > > We defined a gateway in conf/sip_profiles/external/patton.xml: > --------------------- > > --> > > > > > > --------------------- > And the relevant parts of the Patton config: > --------------------- > # define auth > authentication-service AUTH_SVC > username 101 password 101 > > # patton registers > location-service LOCATION_SVC > domain 1 10.156.10.93 > identity 101 > authentication outbound > authenticate 1 authentication-service AUTH_SVC username 101 > registration outbound > registrar 10.156.10.93 5080 > lifetime 3600 > register auto > > context sip-gateway GW_SIP > interface IF_SIP > bind interface IF_IP_WAN context router port 5060 > > context sip-gateway GW_SIP > bind location-service LOCATION_SVC > no shutdown > --------------------- > Output from 'sofia status' > --------------------- > 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG > Name Type Data State > > ================================================================================================= > internal profile sip:mod_sofia at 10.156.10.93:5060 RUNNING > (0) > external profile sip:mod_sofia at 10.156.10.93:5080 RUNNING > (0) > example.com gateway sip:joeuser at example.com NOREG > 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG > 10.156.10.93 alias internal ALIASED > > ================================================================================================= > --------------------- > sip trace > --------------------- > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP 10.156.10.90:5060;branch=z9hG4bKe231fe7b97aa2c0cf > From: ;tag=b793adfadf > To: ;tag=Ua16yjtFgDQ5c > Call-ID: add8fbd86264310b > CSeq: 12517 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > *Before printing this e-mail, please consider the impact on the > environment.* > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/67ad02ee/attachment.html From freeswitch at aastral.net Wed Dec 30 20:06:09 2009 From: freeswitch at aastral.net (Bill W.) Date: Wed, 30 Dec 2009 23:06:09 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <4B3C2331.4030504@aastral.net> Hey VM, Couldn't you just have your core use ODBC instead of SQLite? http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Bill W VM Knott wrote: > I am attempting to use multiple FreeSWITCH servers to share a common > collection of voicemail boxes. Everything is working great, except > for when it comes to managing the custom greetings within the > individual voicemail boxes. > > As I understand it, when creating a custom greeting within a voicemail > box, a record is stored in the voicemail_default.db file located in > the ../freeswitch/db directory (which allows FS to know where to > retrieve the greeting message). My original plan was to link (NFS) > the remote servers to the directory to reference this file, and keep > the custom greetings in an additional shared directory location > (again, all of this working great so far). > > voicemail_prefs cols: username, domain, name_path, greeting_path, password > > Unfortunately, the record that is inserted into this file includes an > IP address (domain) of the FS server handling the call. This > complicates things if I am ?hot swapping? FS servers in and out of the > server cluster. Each time I add a server to the cluster, I will have > to have a process go into this file and replicate all of the records > for each mailbox account, to account for the specific server ip > address. Seems awkward. > > Am I approaching this wrong? i.e., is there a way to by pass this > voicemail_prefs table? > or would I have to dig into the mod_voicemail.c source code and customize it? > > - VMK > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Wed Dec 30 21:53:58 2009 From: freeswitch at aastral.net (Bill W.) Date: Thu, 31 Dec 2009 00:53:58 -0500 Subject: [Freeswitch-users] Sofia sqlite error Message-ID: <4B3C3C76.10007@aastral.net> Hi All I have 3 sofia profiles. Two of them are using sqlite, and one is using odbc. Whenever I restart a profile that is using sqlite, I get the following error: 2009-12-31 00:29:56.914495 [ERR] switch_core_db.c:109 SQL ERR [table sip_shared_appearance_dialogs has no column named network_ip] I'm on version FreeSWITCH Version 1.0.trunk (16055) I did a checkout from trunk and switch_core_db.c hasn't changed from the version I have, so I didn't bother to recompile. Thoughts? Thanks! Bill From pmhshz at gmail.com Wed Dec 30 22:49:48 2009 From: pmhshz at gmail.com (shehzad p) Date: Wed, 30 Dec 2009 22:49:48 -0800 (PST) Subject: [Freeswitch-users] Re cording call into existing file Message-ID: <26975973.post@talk.nabble.com> Hi, while recording a file using session_record, can i continue the existing recorded file? So that the existing record will remain as it is and new recording will be added into that file? Thanks msp -- View this message in context: http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tzury.by at reguluslabs.com Wed Dec 30 22:59:30 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 31 Dec 2009 08:59:30 +0200 Subject: [Freeswitch-users] Configure FS to reply with domain instead of with IP to: user@domain instead of to: user@ip:port Message-ID: <10128ef10912302259o60b989ecxd2a5958a24d98b2c@mail.gmail.com> Hi all, I would like to know what should be done in order to have FreeSWITCH reply with domains instead of IP-Addresses. In my server, DNS NAPTR & SRV Records are all set and working fine. Clients are connected using our pjsip port for windows mobile where transport=tls. When we connect to a public SIP service (e.g. iptel.org) we get the domain when performing INVITE request. We want to gain this behavior in FS as well so it will reply similarly. I am referring to the "TO" part in the message e.g. To: sip:user at domain instead of To: see below our client log in both cases (iptel and FS) thanks in advance for your help, Tzury ## IPTel incoming invite recording DEBUG: 01:26:01.088 pjsua_core.c RX 1746 bytes Request msg INVITE/cseq=7942 (rdata0046B234) from UDP 213.192.59.75:5060: INVITE sip:tal2 at 95.35.38.192:5060;transport=UDP SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 213.192.59.75;branch=z9hG4bKd3a7.502cfbb.0;i=12c8 Via: SIP/2.0/tcp 10.0.0.103:3600;received=80.74.97.189;rport=3600;branch=z9hG4bKPj868b3031276d4f0d9f2338b854b63191 Max-Forwards: 16 From: sip:Tal3 at iptel.org;tag=ff25112de1c843a58df49f57c54a29a2 To: sip:tal2 at iptel.org Contact: ### Freeswitch incoming invite recording DEBUG: 01:28:39.163 pjsua_core.c RX 1257 bytes Request msg INVITE/cseq=119747894 (rdata004B8384) from UDP 67.23.5.142:5060: INVITE sip:1002 at 95.35.115.158:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKyme7e23ggp2QB Max-Forwards: 69 From: "Extension 1000" ;tag=6F8v2e9NNgZ6a To: Call-ID: a9e42c07-10e5-122d-de9e-40402384297d CSeq: 119747894 INVITE Contact: From jcasale at activenetwerx.com Thu Dec 31 00:17:28 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 08:17:28 +0000 Subject: [Freeswitch-users] Outbound Dial Configuration Message-ID: I am starting to migrate an Asterisk box over with a tdm card w/ 1 fxo port this office uses for redundancy when either their sip provider or net connection drops. In Asterisk, I had a long macro for attempting the sip providers first, then finally getting to the dahdi line. Can I simply do the following: to attempt my sip provider first always, then hit span 1/port 1 of my tdm card if it's the provider is not available? Is this an elegant enough way to do this for an office of < 10 phones? Thanks, jlc From jcasale at activenetwerx.com Thu Dec 31 00:45:27 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 08:45:27 +0000 Subject: [Freeswitch-users] Zap problems Message-ID: I am trying to setup a TDM410P card nuder freeswitch-1.0.5-20091230-0400 and fs is reporting it as down? /etc/dahdi/system.conf as it was under Asterisk, so its correct (set to fxs). /opt/freeswitch/conf/zt.conf is default. # cat /opt/freeswitch/conf/openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 # cat /opt/freeswitch/conf/autoload_configs/openzap.conf.xml # dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM410P Board 1 name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM410P location=PCI Bus 10 Slot 02 basechan=1 totchans=4 irq=233 type=analog port=1,FXO port=2,none port=3,none port=4,none Yet fs_cli reports: freeswitch at 127.0.0.1@internal> oz list +OK span: 1 (span1) type: analog chan_count: 1 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options 3way freeswitch at 127.0.0.1@internal> oz dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: FXO state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE Any idea what I am missing? The udev rules are set to the user/group fs runs under. Starting fs at the cli shows the following output, which looks fine? 2009-12-31 01:39:17.592736 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/openzap.conf. 2009-12-31 01:39:17.592803 [DEBUG] zap_io.c:2362 found config for span 2009-12-31 01:39:17.593066 [NOTICE] ozmod_zt.c:1166 Using DAHDI control device 2009-12-31 01:39:17.593112 [INFO] zap_io.c:2579 Loading IO from /opt/freeswitch/mod/ozmod_zt.so [zt] 2009-12-31 01:39:17.593145 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/zt.conf. 2009-12-31 01:39:17.593258 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000 2009-12-31 01:39:17.593298 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000 2009-12-31 01:39:17.593353 [INFO] zap_io.c:2379 auto-loaded 'zt' 2009-12-31 01:39:17.593399 [DEBUG] zap_io.c:2400 created span 1 (span1) of type zt 2009-12-31 01:39:17.593426 [DEBUG] zap_io.c:2413 span 1 [name]=[OpenZAP] 2009-12-31 01:39:17.593457 [DEBUG] zap_io.c:2413 span 1 [number]=[1] 2009-12-31 01:39:17.593485 [DEBUG] zap_io.c:2413 span 1 [fxo-channel]=[1] 2009-12-31 01:39:17.593513 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-31 01:39:17.593593 [INFO] ozmod_zt.c:385 configuring device /dev/dahdi/channel channel 1 as OpenZAP device 1:1 fd:35 2009-12-31 01:39:17.593657 [INFO] zap_io.c:2502 Configured 1 channel(s) although stopping fs yielded this: 2009-12-31 01:42:44.545768 [INFO] zap_io.c:269 Closing channel zt:1:1 fd:35 2009-12-31 01:42:44.574577 [ERR] ozmod_analog.c:953 Failure Polling event! [no matching descriptor] 2009-12-31 01:42:45.546544 [INFO] zap_io.c:2694 Unloading /opt/freeswitch/mod/ozmod_analog.so 2009-12-31 01:42:45.546735 [INFO] zap_io.c:2679 Unloading IO zt 2009-12-31 01:42:45.546756 [INFO] zap_io.c:2694 Unloading /opt/freeswitch/mod/ozmod_zt.so Thanks! jlc From codecomplete at free.fr Thu Dec 31 04:52:14 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 31 Dec 2009 04:52:14 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> References: <26808383.post@talk.nabble.com> <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Message-ID: <26977788.post@talk.nabble.com> Hristo Benev-3 wrote: > Just for your information there is a version of nmap for windows. So you > can do the test from your desktop. Thanks for the info but I'd like to run a test from the Net to see if the NAT firewall did open the required ports for SIP/RTP. I don't have a second PC elsewhere on the Net where I could aim nmap at my home ADSL modem/router. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26977788.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Thu Dec 31 06:14:34 2009 From: jaybinks at gmail.com (Jay Binks) Date: Fri, 1 Jan 2010 00:14:34 +1000 Subject: [Freeswitch-users] Happy new year all Message-ID: Happy new years... 2010 a great new year for FS, looking forward to cluecon already ;) Jay From anthony.minessale at gmail.com Thu Dec 31 07:04:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:04:17 -0600 Subject: [Freeswitch-users] Re cording call into existing file In-Reply-To: <26975973.post@talk.nabble.com> References: <26975973.post@talk.nabble.com> Message-ID: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> set RECORD_APPEND=true on the channel and all recordings will behave this way to formats which support it (curently mod_sndfile for WAV etc) On Thu, Dec 31, 2009 at 12:49 AM, shehzad p wrote: > > Hi, > > while recording a file using session_record, can i continue the existing > recorded file? So that the existing record will remain as it is and new > recording will be added into that file? > > Thanks > msp > -- > View this message in context: > http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/72f665c9/attachment.html From anthony.minessale at gmail.com Thu Dec 31 07:07:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:07:04 -0600 Subject: [Freeswitch-users] Happy new year all In-Reply-To: References: Message-ID: <191c3a030912310707q2f72b63dg10f956ecc54df19f@mail.gmail.com> Thanks for the head's up! It's nice to have eyes into the future. Especially 10 years ago in the eve of y2k =p On Thu, Dec 31, 2009 at 8:14 AM, Jay Binks wrote: > Happy new years... > > 2010 a great new year for FS, looking forward to cluecon already ;) > > Jay > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/5ce8d3d9/attachment-0001.html From rupa at rupa.com Thu Dec 31 07:07:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 31 Dec 2009 09:07:58 -0600 Subject: [Freeswitch-users] Outbound Dial Configuration In-Reply-To: References: Message-ID: Yes. On Thu, Dec 31, 2009 at 2:17 AM, Joseph L. Casale wrote: > I am starting to migrate an Asterisk box over with a tdm card w/ 1 fxo port this > office uses for redundancy when either their sip provider or net connection drops. > > In Asterisk, I had a long macro for attempting the sip providers first, then finally > getting to the dahdi line. > > Can I simply do the following: > > > > to attempt my sip provider first always, then hit span 1/port 1 of my tdm card if it's > the provider is not available? Is this an elegant enough way to do this for an office > of < 10 phones? > > Thanks, > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From saigop at gmail.com Thu Dec 31 07:11:14 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 31 Dec 2009 20:41:14 +0530 Subject: [Freeswitch-users] Module for TTS Message-ID: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> Hi, I would like to develop a module for TTS using http://sourceforge.net/projects/dhvani/ for Freeswitch, How to start with it? -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/eef48098/attachment.html From brian at freeswitch.org Thu Dec 31 07:21:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 31 Dec 2009 09:21:06 -0600 Subject: [Freeswitch-users] Module for TTS In-Reply-To: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> References: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> Message-ID: You might want to take a look at mod_tts_commandline.c as that tts engine is GPL which is license incompatible with FreeSWITH unless they provide an exception. Thanks, Brian On Dec 31, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > Hi, > > I would like to develop a module for TTS using http://sourceforge.net/projects/dhvani/ for Freeswitch, How to start with it? > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/3896a974/attachment.html From anthony.minessale at gmail.com Thu Dec 31 07:32:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:32:29 -0600 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <4B3C3C76.10007@aastral.net> References: <4B3C3C76.10007@aastral.net> Message-ID: <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> delete all the files in your /usr/local/freeswitch/db and rebuild trunk and try again. On Wed, Dec 30, 2009 at 11:53 PM, Bill W. wrote: > Hi All > > I have 3 sofia profiles. Two of them are using sqlite, and one is using > odbc. > > Whenever I restart a profile that is using sqlite, I get the following > error: > > 2009-12-31 00:29:56.914495 [ERR] switch_core_db.c:109 SQL ERR [table > sip_shared_appearance_dialogs has no column named network_ip] > > I'm on version FreeSWITCH Version 1.0.trunk (16055) > > I did a checkout from trunk and switch_core_db.c hasn't changed from the > version I have, so I didn't bother to recompile. > > Thoughts? > > Thanks! > Bill > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/c4b9338b/attachment.html From freeswitch at peely.com Thu Dec 31 07:49:50 2009 From: freeswitch at peely.com (peely) Date: Thu, 31 Dec 2009 07:49:50 -0800 (PST) Subject: [Freeswitch-users] Call through gateway without register > sends to gateway name? Message-ID: <26979541.post@talk.nabble.com> Hi, I have a problem where I'm trying to send calls to a gateway that does not support registration. In my external sip profile directory I have a file containing: Then in my dialplan I have a dialplan with an action of: However, when I call out, the Sofia diag shows: sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR mygateway (to [172.16.1.1]:53) Could somebody please tell me how I get the gateway config to send INVITEs to a specific IP? Thanks, Neil. -- View this message in context: http://old.nabble.com/Call-through-gateway-without-register-%3E-sends-to-gateway-name--tp26979541p26979541.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Dec 31 08:00:05 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 31 Dec 2009 08:00:05 -0800 Subject: [Freeswitch-users] Zap problems In-Reply-To: References: Message-ID: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> Actually DOWN simply means idle. Go ahead and make some calls - your config looks okay. -MC On Dec 31, 2009, at 12:45 AM, "Joseph L. Casale" wrote: > I am trying to setup a TDM410P card nuder > freeswitch-1.0.5-20091230-0400 > and fs is reporting it as down? > > /etc/dahdi/system.conf as it was under Asterisk, so its correct (set > to fxs). > /opt/freeswitch/conf/zt.conf is default. > > # cat /opt/freeswitch/conf/openzap.conf > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > # cat /opt/freeswitch/conf/autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > > > > > > # dahdi_scan > [1] > active=yes > alarms=OK > description=Wildcard TDM410P Board 1 > name=WCTDM/0 > manufacturer=Digium > devicetype=Wildcard TDM410P > location=PCI Bus 10 Slot 02 > basechan=1 > totchans=4 > irq=233 > type=analog > port=1,FXO > port=2,none > port=3,none > port=4,none > > Yet fs_cli reports: > > freeswitch at 127.0.0.1@internal> oz list > +OK > span: 1 (span1) > type: analog > chan_count: 1 > dialplan: XML > context: default > dial_regex: > fail_dial_regex: > hold_music: > analog_options 3way > > freeswitch at 127.0.0.1@internal> oz dump 1 1 > span_id: 1 > chan_id: 1 > physical_span_id: 1 > physical_chan_id: 1 > type: FXO > state: DOWN > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > Any idea what I am missing? The udev rules are set to the user/group > fs runs under. > Starting fs at the cli shows the following output, which looks fine? > > 2009-12-31 01:39:17.592736 [DEBUG] zap_config.c:56 Configuration > file is /opt/freeswitch/conf/openzap.conf. > 2009-12-31 01:39:17.592803 [DEBUG] zap_io.c:2362 found config for span > 2009-12-31 01:39:17.593066 [NOTICE] ozmod_zt.c:1166 Using DAHDI > control device > 2009-12-31 01:39:17.593112 [INFO] zap_io.c:2579 Loading IO from /opt/ > freeswitch/mod/ozmod_zt.so [zt] > 2009-12-31 01:39:17.593145 [DEBUG] zap_config.c:56 Configuration > file is /opt/freeswitch/conf/zt.conf. > 2009-12-31 01:39:17.593258 [INFO] ozmod_zt.c:556 Setting rxgain val > to 0.000000 > 2009-12-31 01:39:17.593298 [INFO] ozmod_zt.c:565 Setting txgain val > to 0.000000 > 2009-12-31 01:39:17.593353 [INFO] zap_io.c:2379 auto-loaded 'zt' > 2009-12-31 01:39:17.593399 [DEBUG] zap_io.c:2400 created span 1 > (span1) of type zt > 2009-12-31 01:39:17.593426 [DEBUG] zap_io.c:2413 span 1 [name]= > [OpenZAP] > 2009-12-31 01:39:17.593457 [DEBUG] zap_io.c:2413 span 1 [number]=[1] > 2009-12-31 01:39:17.593485 [DEBUG] zap_io.c:2413 span 1 [fxo-channel] > =[1] > 2009-12-31 01:39:17.593513 [DEBUG] zap_io.c:2442 setting trunk type > to 'FXO' start(KEWL) > 2009-12-31 01:39:17.593593 [INFO] ozmod_zt.c:385 configuring device / > dev/dahdi/channel channel 1 as OpenZAP device 1:1 fd:35 > 2009-12-31 01:39:17.593657 [INFO] zap_io.c:2502 Configured 1 channel > (s) > > although stopping fs yielded this: > > 2009-12-31 01:42:44.545768 [INFO] zap_io.c:269 Closing channel zt: > 1:1 fd:35 > 2009-12-31 01:42:44.574577 [ERR] ozmod_analog.c:953 Failure Polling > event! [no matching descriptor] > 2009-12-31 01:42:45.546544 [INFO] zap_io.c:2694 Unloading /opt/ > freeswitch/mod/ozmod_analog.so > 2009-12-31 01:42:45.546735 [INFO] zap_io.c:2679 Unloading IO zt > 2009-12-31 01:42:45.546756 [INFO] zap_io.c:2694 Unloading /opt/ > freeswitch/mod/ozmod_zt.so > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Dec 31 08:07:19 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 31 Dec 2009 10:07:19 -0600 Subject: [Freeswitch-users] Call through gateway without register > sends to gateway name? In-Reply-To: <26979541.post@talk.nabble.com> References: <26979541.post@talk.nabble.com> Message-ID: <04A52674-4ABB-4EAB-BFA9-65A18C3700C5@freeswitch.org> If you don't call your gateway the dns name or the IP then you'll have to specify the proxy/username/password/from-domain settings. /b On Dec 31, 2009, at 9:49 AM, peely wrote: > > > From jcasale at activenetwerx.com Thu Dec 31 08:16:12 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 16:16:12 +0000 Subject: [Freeswitch-users] Zap problems In-Reply-To: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> References: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> Message-ID: >Actually DOWN simply means idle. Go ahead and make some calls - your >config looks okay. Heh, I had the guys onsite re-install the old discs as I figured I couldn't get the thing operational by start of business... I'll be back on it once I recover from tonight:) Thanks for all the help guys, happy new year! jlc From jcasale at activenetwerx.com Thu Dec 31 08:19:04 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 16:19:04 +0000 Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <26977788.post@talk.nabble.com> References: <26808383.post@talk.nabble.com> <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <26977788.post@talk.nabble.com> Message-ID: >Thanks for the info but I'd like to run a test from the Net to see if the >NAT firewall did open the required ports for SIP/RTP. I don't have a second >PC elsewhere on the Net where I could aim nmap at my home ADSL modem/router. http://www.t1shopper.com/tools/port-scanner/ One of the few that allow you to scan an ip that's not yours... From msc at freeswitch.org Thu Dec 31 09:20:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Dec 2009 09:20:14 -0800 Subject: [Freeswitch-users] freeswitch and H323 In-Reply-To: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> Message-ID: <87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> Are you trying to use mod_h323 or mod_opal? They are both works in progress, but the latter is farther along than the former. Use the latest FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh script in the build directory. If you have any build issues then paste the log on pastebin.freeswitch.org and reply to this thread with the PB URL so that we can take a look. -MC On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay wrote: > Hi, > > has anyone been able to get H323 to work? > > I have problem trying to get it compiled with either 1.0.4 or 1.0.5. > > Thanks, > pete > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/b8ee3170/attachment.html From msc at freeswitch.org Thu Dec 31 09:31:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Dec 2009 09:31:07 -0800 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? In-Reply-To: <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> References: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> Message-ID: <87f2f3b90912310931o35387aceq6c7f9cca7726bdbf@mail.gmail.com> On Wed, Dec 30, 2009 at 8:23 AM, Brian West wrote: > Someone could pay to have support added but as it stands we don't have it > yet. If its going to be done it has to be proper mgcp support not half ass > support. > > As Brian noted on yesterday's conf call he occasionally sends "terse" emails. However, being terse frequently gets the point across in no uncertain terms. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/bc43cfab/attachment.html From freeswitch at aastral.net Thu Dec 31 11:07:34 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 31 Dec 2009 14:07:34 -0500 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> References: <4B3C3C76.10007@aastral.net> <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> Message-ID: <4B3CF676.8040908@aastral.net> Worked! Was that a one-time thing or do I need to blow away the sqlite database every time I upgrade? Or more generally, how does FreeSWITCH handle database updates and what should the procedures be to ensure the database remains current and no data is lost by having to remove database files? Thanks! Bill Anthony Minessale wrote: > delete all the files in your /usr/local/freeswitch/db and rebuild trunk > and try again. > > From anthony.minessale at gmail.com Thu Dec 31 12:02:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 14:02:45 -0600 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <4B3CF676.8040908@aastral.net> References: <4B3C3C76.10007@aastral.net> <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> <4B3CF676.8040908@aastral.net> Message-ID: <191c3a030912311202t15d1fa5p4799de5114bb867c@mail.gmail.com> usually it does a test and erases the table and recreates it when need be. On Thu, Dec 31, 2009 at 1:07 PM, Bill W wrote: > Worked! > > Was that a one-time thing or do I need to blow away the sqlite database > every time I upgrade? > > Or more generally, how does FreeSWITCH handle database updates and what > should the procedures be to ensure the database remains current and no > data is lost by having to remove database files? > > Thanks! > Bill > > > Anthony Minessale wrote: > > delete all the files in your /usr/local/freeswitch/db and rebuild trunk > > and try again. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/37212650/attachment.html From stevendt at primrosebank.net Thu Dec 31 12:22:06 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 31 Dec 2009 20:22:06 -0000 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does it mean ? Message-ID: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> Hi Guys Happy New Year ! - 4 hours to midnight here - what am I doing looking at the console log ???? Anyway, all the best for 2010, and when someone gets a chance to look at this, could someone please help me understand what the following error message means ? I've seen it a few times before and not seen anything untoward. My "production" system is running 1.0.4 - reported as (14460) but it is somewhat later than that (15xxx). I plan on doing a new SVN build over the holiday, but don't expect this error to go away ? Message is along the lines . . . . date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/87c358d9/attachment.html From anthony.minessale at gmail.com Thu Dec 31 12:33:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 14:33:46 -0600 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does it mean ? In-Reply-To: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> Message-ID: <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> It means you have a profound jump in time like someone set the clock to an earlier time and it's recovering by re-syncing the clock. You can avoid this by using a monotonic clock which must not be possible on your system because it uses it by default. What OS is it? On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson wrote: > Hi Guys > > Happy New Year ! > > - 4 hours to midnight here - what am I doing looking at the console log > ???? > > > Anyway, all the best for 2010, and when someone gets a chance to look at > this, could someone please help me understand what the following error > message means ? > > I've seen it a few times before and not seen anything untoward. My > "production" system is running 1.0.4 - reported as (14460) but it is > somewhat later than that (15xxx). I plan on doing a new SVN build over the > holiday, but don't expect this error to go away ? > > Message is along the lines . . . . > > > date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/f10f4972/attachment.html From vmknott at gmail.com Thu Dec 31 14:03:57 2009 From: vmknott at gmail.com (VM Knott) Date: Thu, 31 Dec 2009 17:03:57 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) Message-ID: Thank you Bill for the input, but changing how I access the data source does not solve my problem. I was hoping to avoid the management of IP Addresses for every voicemail box on the system. Is there a way for me to set a default greeting to all voicemail boxes globally, without having to go to a repository (regardless of means of access) for each mailbox? ---------- Forwarded message ---------- From: "Bill W." To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Dec 2009 23:06:09 -0500 Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) Hey VM, Couldn't you just have your core use ODBC instead of SQLite? http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Bill W VM Knott wrote: > I am attempting to use multiple FreeSWITCH servers to share a common > collection of voicemail boxes. Everything is working great, except > for when it comes to managing the custom greetings within the > individual voicemail boxes. > > As I understand it, when creating a custom greeting within a voicemail > box, a record is stored in the voicemail_default.db file located in > the ../freeswitch/db directory (which allows FS to know where to > retrieve the greeting message). My original plan was to link (NFS) > the remote servers to the directory to reference this file, and keep > the custom greetings in an additional shared directory location > (again, all of this working great so far). > > voicemail_prefs cols: username, domain, name_path, greeting_path, password > > Unfortunately, the record that is inserted into this file includes an > IP address (domain) of the FS server handling the call. This > complicates things if I am ?hot swapping? FS servers in and out of the > server cluster. Each time I add a server to the cluster, I will have > to have a process go into this file and replicate all of the records > for each mailbox account, to account for the specific server ip > address. Seems awkward. > > Am I approaching this wrong? i.e., is there a way to by pass this > voicemail_prefs table? > or would I have to dig into the mod_voicemail.c source code and customize it? > > - VMK > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 31 14:19:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 16:19:03 -0600 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <191c3a030912311419h3540a684q2f259aa8687f04a4@mail.gmail.com> use a real domain instead of IP might work. On Thu, Dec 31, 2009 at 4:03 PM, VM Knott wrote: > Thank you Bill for the input, but changing how I access the data > source does not solve my problem. > I was hoping to avoid the management of IP Addresses for every > voicemail box on the system. > > Is there a way for me to set a default greeting to all voicemail boxes > globally, without having to go to a repository (regardless of means of > access) for each mailbox? > > > > ---------- Forwarded message ---------- > From: "Bill W." > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Dec 2009 23:06:09 -0500 > Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) > Hey VM, > > Couldn't you just have your core use ODBC instead of SQLite? > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > -Bill W > > > VM Knott wrote: > > I am attempting to use multiple FreeSWITCH servers to share a common > > collection of voicemail boxes. Everything is working great, except > > for when it comes to managing the custom greetings within the > > individual voicemail boxes. > > > > As I understand it, when creating a custom greeting within a voicemail > > box, a record is stored in the voicemail_default.db file located in > > the ../freeswitch/db directory (which allows FS to know where to > > retrieve the greeting message). My original plan was to link (NFS) > > the remote servers to the directory to reference this file, and keep > > the custom greetings in an additional shared directory location > > (again, all of this working great so far). > > > > voicemail_prefs cols: username, domain, name_path, greeting_path, > password > > > > Unfortunately, the record that is inserted into this file includes an > > IP address (domain) of the FS server handling the call. This > > complicates things if I am ?hot swapping? FS servers in and out of the > > server cluster. Each time I add a server to the cluster, I will have > > to have a process go into this file and replicate all of the records > > for each mailbox account, to account for the specific server ip > > address. Seems awkward. > > > > Am I approaching this wrong? i.e., is there a way to by pass this > > voicemail_prefs table? > > or would I have to dig into the mod_voicemail.c source code and customize > it? > > > > - VMK > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/94d6d7e1/attachment.html From stevendt at primrosebank.net Thu Dec 31 15:02:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 31 Dec 2009 23:02:23 -0000 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> Message-ID: Hi Anthony it's Windows XP - there was no time change on the PC any time close to this event - or indeed for weeks since FreeSwitch was installed. I'll check, but I don't think the machine is even using SNTP. What does FreeSwitch define as a "profound jump in time" ? regards Dave (off to see the New Year in now !) ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 31, 2009 8:33 PM Subject: Re: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? It means you have a profound jump in time like someone set the clock to an earlier time and it's recovering by re-syncing the clock. You can avoid this by using a monotonic clock which must not be possible on your system because it uses it by default. What OS is it? On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson wrote: Hi Guys Happy New Year ! - 4 hours to midnight here - what am I doing looking at the console log ???? Anyway, all the best for 2010, and when someone gets a chance to look at this, could someone please help me understand what the following error message means ? I've seen it a few times before and not seen anything untoward. My "production" system is running 1.0.4 - reported as (14460) but it is somewhat later than that (15xxx). I plan on doing a new SVN build over the holiday, but don't expect this error to go away ? Message is along the lines . . . . date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/655bb922/attachment.html From rupa at rupa.com Thu Dec 31 15:56:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 31 Dec 2009 17:56:24 -0600 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? In-Reply-To: References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> Message-ID: Don't most modern windows operating systems automatically sync with time.windows.com or something like that? On Thu, Dec 31, 2009 at 5:02 PM, Dave Stevenson wrote: > Hi Anthony > > it's Windows XP - there was no time change on the PC any time close to this > event - or indeed for weeks since FreeSwitch was installed. > > I'll check, but I don't think the machine is even using SNTP. > > What does FreeSwitch define as a "profound jump in time" ? > > regards > Dave > > (off to see the New Year in now !) > > ----- Original Message ----- > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, December 31, 2009 8:33 PM > *Subject:* Re: [Freeswitch-users] "Reverse Clock Skew Detected" - what > does itmean ? > > It means you have a profound jump in time like someone set the clock to an > earlier time and it's recovering by re-syncing the clock. > You can avoid this by using a monotonic clock which must not be possible on > your system because it uses it by default. > > What OS is it? > > > On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson > wrote: > >> Hi Guys >> >> Happy New Year ! >> >> - 4 hours to midnight here - what am I doing looking at the console log >> ???? >> >> >> Anyway, all the best for 2010, and when someone gets a chance to look at >> this, could someone please help me understand what the following error >> message means ? >> >> I've seen it a few times before and not seen anything untoward. My >> "production" system is running 1.0.4 - reported as (14460) but it is >> somewhat later than that (15xxx). I plan on doing a new SVN build over the >> holiday, but don't expect this error to go away ? >> >> Message is along the lines . . . . >> >> >> date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" >> >> regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/9e3e159a/attachment-0001.html From sharad at coraltele.com Thu Dec 31 22:20:47 2009 From: sharad at coraltele.com (Sharad) Date: Thu, 31 Dec 2009 22:20:47 -0800 (PST) Subject: [Freeswitch-users] Self alarm In-Reply-To: <1262250725607-4235713.post@n2.nabble.com> References: <1262250725607-4235713.post@n2.nabble.com> Message-ID: <1262326847726-4238924.post@n2.nabble.com> Hi I am also intresting in the same. Is there any script for this functionality. Regards -- View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 31 22:31:14 2009 From: freeswitch at aastral.net (Bill W.) Date: Fri, 01 Jan 2010 01:31:14 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <4B3D96B2.4060402@aastral.net> So is the problem that you're having to replicate the voicemail database across switches in the cluster or is the problem the content of the entries in voicemail database? Because in your original post you're speaking of trying to share the voicemail db over NFS. Thanks, Bill VM Knott wrote: > Thank you Bill for the input, but changing how I access the data > source does not solve my problem. > I was hoping to avoid the management of IP Addresses for every > voicemail box on the system. > > Is there a way for me to set a default greeting to all voicemail boxes > globally, without having to go to a repository (regardless of means of > access) for each mailbox? > > > From freeswitch at aastral.net Thu Dec 31 23:29:46 2009 From: freeswitch at aastral.net (Bill W.) Date: Fri, 01 Jan 2010 02:29:46 -0500 Subject: [Freeswitch-users] Re cording call into existing file In-Reply-To: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> References: <26975973.post@talk.nabble.com> <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> Message-ID: <4B3DA46A.4010906@aastral.net> Added to the Wiki. http://wiki.freeswitch.org/wiki/Variable_RECORD_APPEND Anthony Minessale wrote: > set RECORD_APPEND=true on the channel and all recordings will behave > this way to formats which support it > (curently mod_sndfile for WAV etc) > From devel at thom.fr.eu.org Tue Dec 1 00:37:59 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Tue, 01 Dec 2009 09:37:59 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= Message-ID: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale at hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [12] iax:guest at conference.freeswitch.org/888 [13] googletalk:conf+888 at conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:FreeSWITCH-users at lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale at hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale at gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888 at conference.freeswitch.org [13] http://iax:guest at conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/06045a86/attachment-0002.html From dome at tel.co.th Tue Dec 1 05:43:06 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 1 Dec 2009 20:43:06 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 Message-ID: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Dear All, I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and freeswitch 1.0.5 pre 7 My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and 2 GB memmory. when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% when i disable echo cancel FS can start but when first call incominng got 100% again. Someone can help me? BG Dome -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/5082ddfe/attachment-0002.html From juanbackson at gmail.com Tue Dec 1 05:59:42 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 1 Dec 2009 21:59:42 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true Message-ID: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> Hi, I found that with bypass_media=true, freeswitch would change c= to FS's own IP. I think this is a misconfiguration. Does anyone know what config could have caused that? thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/30430685/attachment-0002.html From juanbackson at gmail.com Tue Dec 1 07:08:12 2009 From: juanbackson at gmail.com (Juan Backson) Date: Tue, 1 Dec 2009 23:08:12 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> Message-ID: <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> In the following trace, 102 is FS IP, 104 is calling party and 13 is called party. with bypass_media, FS still changes c=IN IP4 192.168.1.102 Any idea why? freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 22:56:33.782715: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport From: >;tag=786224322 To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 CSeq: 37 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 397 v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 From: >;tag=786224322 To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 CSeq: 37 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ PhonerLite at 192.168.1.102 entering state [received][100] 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259708193783162] Caller-Channel-Created-Time: [1259708193783162] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [786224322] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_bypass_media: [true] variable_proxy_media: [true] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 3393406017 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ 90964111 at 192.168.1.116:9390 SOFIA ROUTING 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g Max-Forwards: 69 From: "PhonerLite" >;tag=8tH6Xjt2XaU9F To: Call-ID: 9d052856-596f-122d-1b98-0022190e9476 CSeq: 123735760 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 404 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/56510807/attachment-0002.html From moises.silva at gmail.com Tue Dec 1 08:32:04 2009 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 1 Dec 2009 11:32:04 -0500 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Message-ID: On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: > Dear All, > I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I > use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and > freeswitch 1.0.5 pre 7 > > My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and > 2 GB memmory. > when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% > when i disable echo cancel FS can start but when first call incominng got > 100% again. > > > Someone can help me? > > > I suggest you to start using freeswitch trunk (and openzap trunk). In the other hand, when you talk about 4E1 R2, you mean R2 as the telephony signaling or is this some kind of board brand? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1dae8f4e/attachment-0002.html From mike at jerris.com Tue Dec 1 08:42:44 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 11:42:44 -0500 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> Message-ID: <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> What is the jira bug number on this voicemail email issue? I don't recall seeing it. Mike On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine wrote: > > Are you on SVN trunk? As far as I recall the callee_id_number/name > stuff isnt in 1.0.4. > > No, because the SVN has problems with Emailing the voicemail... > > We use 1.0.4 and set sip_callee_id_number/name which works. I would > like to not set it and get it from the other side... > > Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 1 08:46:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 11:46:08 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> Message-ID: The only way this would happen would be if this is set to proxy media not bypass. Are you setting both? Mike On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > In the following trace, 102 is FS IP, 104 is calling party and 13 > is called party. > > with bypass_media, FS still changes c=IN IP4 192.168.1.102 > > Any idea why? > > > freeswitch at localhost.localdomain> recv 951 bytes from udp/ > [192.168.1.104]:5060 at 22:56:33.782715: > > --- > --------------------------------------------------------------------- > INVITE sip:90964111 at 192.168.1.102 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport > From: ;tag=786224322 > To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > Contact: > Content-Type: application/sdp > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE > Max-Forwards: 70 > Supported: 100rel, replaces > User-Agent: SIPPER for PhonerLite > Content-Length: 397 > > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendrecv > > --- > --------------------------------------------------------------------- > send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: > > --- > --------------------------------------------------------------------- > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 > From: ;tag=786224322 > To: > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Content-Length: 0 > > > --- > --------------------------------------------------------------------- > 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a- > ee3b-40d4-910d-3b1579f9a273] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ > internal/PhonerLite at 192.168.1.102 entering state [received][100] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > > EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() > 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259708193783162] > Caller-Channel-Created-Time: [1259708193783162] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [786224322] > variable_sofia_profile_name: [internal] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] > variable_sip_call_id: [003C8E1B-F8DC-DE11- > A853-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_bypass_media: [true] > variable_proxy_media: [true] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_endpoint_disposition: [RECEIVED_NOMEDIA] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > > > 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ > 90964111 at 192.168.1.116:9390 Patched SDP > --- > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > +++ > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send > signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 > (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ > 90964111 at 192.168.1.116:9390 SOFIA ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/ > internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send > signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to > sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > going to sleep > send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: > > --- > --------------------------------------------------------------------- > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g > Max-Forwards: 69 > From: "PhonerLite" ;tag=8tH6Xjt2XaU9F > To: > Call-ID: 9d052856-596f-122d-1b98-0022190e9476 > CSeq: 123735760 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 404 > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off > > v=0 > o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/92126985/attachment-0002.html From dome at tel.co.th Tue Dec 1 09:02:18 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Dec 2009 00:02:18 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> Message-ID: <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> 2009/12/1 Moises Silva > On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: > >> Dear All, >> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). >> I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and >> freeswitch 1.0.5 pre 7 >> >> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz and >> 2 GB memmory. >> when i start freeswitch if enable echo cencel in zt.conf CPU peak to 100% >> when i disable echo cancel FS can start but when first call incominng got >> 100% again. >> >> >> Someone can help me? >> >> >> > I suggest you to start using freeswitch trunk (and openzap trunk). > > I'll try > In the other hand, when you talk about 4E1 R2, you mean R2 as the telephony > signaling or is this some kind of board brand? > telephony signaling Thanks. Dome C. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/b5b2b584/attachment-0002.html From yehavi.bourvine at gmail.com Tue Dec 1 09:11:24 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 Dec 2009 19:11:24 +0200 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> Message-ID: It is MODAPP-373. Thanks, __yehavi: 2009/12/1 Michael Jerris > What is the jira bug number on this voicemail email issue? I don't > recall seeing it. > > Mike > > On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine > wrote: > > > > Are you on SVN trunk? As far as I recall the callee_id_number/name > > stuff isnt in 1.0.4. > > > > No, because the SVN has problems with Emailing the voicemail... > > > > We use 1.0.4 and set sip_callee_id_number/name which works. I would > > like to not set it and get it from the other side... > > > > Thanks! __Yehavi: > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/8a2b2ecb/attachment-0002.html From msc at freeswitch.org Tue Dec 1 09:31:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 09:31:47 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: <200911291906.51520.errotan@gmail.com> References: <200911291906.51520.errotan@gmail.com> Message-ID: <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> On Sun, Nov 29, 2009 at 10:06 AM, Pusk?s Zsolt wrote: > Hi Guys! > > I'm using the latest svn (15711) with the default xml config. Only modified > cdr_csv.conf.xml the line to name="legs" > value="ab"/> > > Here is what i do: > > 1. 1000 calls 1001 (1001 answers the call) > 2. 1001 do blind transfer to 1002 (using *1) > 3. 1001 hangs up > 4. 1002 answers the call > 5. 1002 and 1000 hangs up > > 3 cdr records are generated (simplified): > > from,to,start,duration > "1000" "1001" "2009-11-29 15:21:53" "53" "50" > "1000" "1002" "2009-11-29 15:21:53" "79" "76" > "1000" "1002" "2009-11-29 15:22:46" "26" "23" > > As you can see the second cdr is incorrect because 1000 doesn't speak with > 1002 for 76 second. > > Is this a normal ? Is it possible to make only 2 record ? > > You may want to turn on mod_xml_curl and look at XML CDRs, comparing them to the corresponding CSV files. That should help you figure out why the values in the CSV are what they are. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/45551bfa/attachment-0002.html From msc at freeswitch.org Tue Dec 1 09:38:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 09:38:42 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: Message-ID: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> On Wed, Nov 25, 2009 at 9:42 PM, Joseph L. Casale wrote: > I need to make faxing easy for some very computer illiterate folk. I am > using an email > service and going to use procmail to print anything incoming automatically > but they cant > get the hang of scanning to an email app, so I am going to buy a Linksys > PAP2T as per the > wiki. > > Since the setup will never receive inbound remote faxes, I just need to > direct all fax's > sent from the FXS port (that extension) to the email script in the wiki > substituting the > destination # as the alias portion of the email. > > So if I create a dialplan that catches the caller_id_number of the FXS > port, does the $1 > variable exist in the following scenario: > > > > In this case, the $1 will only contain whatever is in the parens in your expression, i.e. What do you have for your expression? -MC > as that's how our service requires fax's, the 10 digit # at their domain, > fax.com. Is this > a plausible setup? > > Lastly, I see in the interop list that Audiocodes Mediapack 114 is > supported, but the 202 > is not listed, is that simply because its new or is it known to not work? > Given that its > the same price as the Linksys, I would rather get it. > > Thanks! > jlc > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/148dec48/attachment-0002.html From ryannyl at gmail.com Tue Dec 1 10:40:26 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 2 Dec 2009 02:40:26 +0800 Subject: [Freeswitch-users] User logon/logout from analog phones Message-ID: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> Dear All: I try to register from a feature code of an analog phone like Elastix. It is useful for DID. There is an idea that I use dynamic dialplan to implement it and it's not really register to FS. And I need to run script to insert or delete dialplan to the database when dialed.(Input logon's ExtNumber and Password) Is that right? or any recommendation? Thanks in advance. -- Sincerely regards, Ryanny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/51a1a06b/attachment-0002.html From msc at freeswitch.org Tue Dec 1 11:04:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 11:04:43 -0800 Subject: [Freeswitch-users] User logon/logout from analog phones In-Reply-To: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> References: <4bfcac7e0912011040k341d3eb5r49da104cd73c9c0d@mail.gmail.com> Message-ID: <87f2f3b90912011104xbbcb86fkc08c9372a34232e6@mail.gmail.com> I'm not entirely sure that I understand your question, so I am going to ask a few questions to clarify. Are you looking to have analog telephones receive incoming calls, like in a call center? Is that why the user of the analog phone would need to log in and log out? I would recommend checkout out mod_xml_curl for the dynamic dialplan. -MC On Tue, Dec 1, 2009 at 10:40 AM, Ryanny Lin wrote: > Dear All: > > I try to register from a feature code of an analog phone like Elastix. > It is useful for DID. > There is an idea that I use dynamic dialplan to implement it and it's not > really register to FS. > And I need to run script to insert or delete dialplan to the database when > dialed.(Input logon's ExtNumber and Password) > Is that right? or any recommendation? > > Thanks in advance. > -- > Sincerely regards, > Ryanny > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/62642775/attachment-0002.html From john_platts at hotmail.com Tue Dec 1 11:19:51 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 1 Dec 2009 13:19:51 -0600 Subject: [Freeswitch-users] Problem with compiling revision 15739 Message-ID: I attempted to do a make current with revision 15739, but some of the Sofia source files will not compile with revision 15739. Those source files were not changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile FreeSWITCH. I used the following to get revision 15738, which was the previous revision, built: make update-clean svn update -r 15738 make all install This does the same stuff as make current, except that revision 15738 is checked out of the SVN repository. _________________________________________________________________ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 From moises.silva at gmail.com Tue Dec 1 11:22:05 2009 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 1 Dec 2009 14:22:05 -0500 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> Message-ID: On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote: > > > 2009/12/1 Moises Silva > > On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: >> >>> Dear All, >>> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). >>> I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) >>> and freeswitch 1.0.5 pre 7 >>> >>> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz >>> and 2 GB memmory. >>> when i start freeswitch if enable echo cencel in zt.conf CPU peak to >>> 100% when i disable echo cancel FS can start but when first call incominng >>> got 100% again. >>> >>> >>> Someone can help me? >>> >>> >>> >> I suggest you to start using freeswitch trunk (and openzap trunk). >> >> > I'll try > > >> In the other hand, when you talk about 4E1 R2, you mean R2 as the >> telephony signaling or is this some kind of board brand? >> > > > telephony signaling > Does the signaling stack comes with the board and is integrated into freeswitch as an endpoint? if not, your only chance is using openr2, which requires freeswitch, openzap and openr2 trunk. http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/23d5c8c2/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 1 11:57:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 13:57:36 -0600 Subject: [Freeswitch-users] Passing incoming remote-party-id from called to caller In-Reply-To: References: <191c3a030911302214u7de314cch45f063e761619041@mail.gmail.com> <191c3a030911302215r15d6e48ha5f2b929f5706829@mail.gmail.com> <6B54A394-BB50-49CA-A299-47CF50AC9540@jerris.com> Message-ID: <191c3a030912011157o1c0e1d3cp37ffe1338373d60b@mail.gmail.com> the updating of the display code is significantly improved in trunk. Please figure out your email problem and use that. Most likely you need an alternate configuration. What mailer client are you using in switch.conf.xml ? On Tue, Dec 1, 2009 at 11:11 AM, Yehavi Bourvine wrote: > It is MODAPP-373. > > Thanks, __yehavi: > > 2009/12/1 Michael Jerris > > What is the jira bug number on this voicemail email issue? I don't >> recall seeing it. >> >> Mike >> >> On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine >> wrote: >> >> > > Are you on SVN trunk? As far as I recall the callee_id_number/name >> > stuff isnt in 1.0.4. >> > >> > No, because the SVN has problems with Emailing the voicemail... >> > >> > We use 1.0.4 and set sip_callee_id_number/name which works. I would >> > like to not set it and get it from the other side... >> > >> > Thanks! __Yehavi: >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/f48b6771/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 1 11:59:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 13:59:09 -0600 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> Message-ID: <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> yes he did you can see it in his trace. you can not use both of them together...... On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > The only way this would happen would be if this is set to proxy media not > bypass. Are you setting both? > > Mike > > On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > > In the following trace, 102 is FS IP, 104 is calling party and 13 is > called party. > > with bypass_media, FS still changes c=IN IP4 192.168.1.102 > > Any idea why? > > > freeswitch at localhost.localdomain> recv 951 bytes from > udp/[192.168.1.104]:5060 at 22:56:33.782715: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.102 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport > From: > >;tag=786224322 > To: > > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > Contact: > Content-Type: application/sdp > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE > Max-Forwards: 70 > Supported: 100rel, replaces > User-Agent: SIPPER for PhonerLite > Content-Length: 397 > > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=sendrecv > ------------------------------------------------------------------------ > send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 > From: > >;tag=786224322 > To: > > Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 > CSeq: 37 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ > PhonerLite at 192.168.1.102 entering state [received][100] > 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > > EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() > 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_EXECUTE] > Channel-State-Number: [4] > Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [ringing] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259708193783162] > Caller-Channel-Created-Time: [1259708193783162] > Caller-Channel-Answered-Time: [0] > Caller-Channel-Progress-Time: [0] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [786224322] > variable_sofia_profile_name: [internal] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] > variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_bypass_media: [true] > variable_proxy_media: [true] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_endpoint_disposition: [RECEIVED_NOMEDIA] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > > > 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ > 90964111 at 192.168.1.116:9390 Patched SDP > --- > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > +++ > v=0 > o=- 3393406017 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ > 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 > (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ > 90964111 at 192.168.1.116:9390 SOFIA ROUTING > 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 > (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA > 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 > (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to > sleep > send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g > Max-Forwards: 69 > From: "PhonerLite" > >;tag=8tH6Xjt2XaU9F > To: > Call-ID: 9d052856-596f-122d-1b98-0022190e9476 > CSeq: 123735760 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 404 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/d2f9cc74/attachment-0002.html From mike at jerris.com Tue Dec 1 12:02:17 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 15:02:17 -0500 Subject: [Freeswitch-users] Problem with compiling revision 15739 In-Reply-To: References: Message-ID: I think I just fixed this a few minutes ago, it is running test builds on the build servers now to verify. On Dec 1, 2009, at 2:19 PM, John Platts wrote: > > I attempted to do a make current with revision 15739, but some of the Sofia source files will not compile with revision 15739. Those source files were not changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile FreeSWITCH. I used the following to get revision 15738, which was the previous revision, built: > make update-clean > svn update -r 15738 > make all install > > This does the same stuff as make current, except that revision 15738 is checked out of the SVN repository. > > _________________________________________________________________ > Windows 7: Unclutter your desktop. Learn more. > http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ryannyl at gmail.com Tue Dec 1 12:27:10 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 2 Dec 2009 04:27:10 +0800 Subject: [Freeswitch-users] User logon/logout from analog phones Message-ID: <4bfcac7e0912011227y58204c84ld0611d50afe6237a@mail.gmail.com> Dear Michael: Yes, I want to distribute a real phone number to each analog phone (direct inward dialing). One FXS one analog phone. I guess the user maybe want to add a number mapping this FXS port. Thank you, Michael. mod_xml_curl is really a powerful module. :D ---------- Forward ---------- From: Michael Collins To: freeswitch-users at lists.freeswitch.org Date: Tue, 1 Dec 2009 11:04:43 -0800 I'm not entirely sure that I understand your question, so I am going to ask a few questions to clarify. Are you looking to have analog telephones receive incoming calls, like in a call center? Is that why the user of the analog phone would need to log in and log out? I would recommend checkout out mod_xml_curl for the dynamic dialplan. -MC On Tue, Dec 1, 2009 at 10:40 AM, Ryanny Lin wrote: > Dear All: > > I try to register from a feature code of an analog phone like Elastix. > It is useful for DID. > There is an idea that I use dynamic dialplan to implement it and it's not > really register to FS. > And I need to run script to insert or delete dialplan to the database when > dialed.(Input logon's ExtNumber and Password) > Is that right? or any recommendation? > > Thanks in advance. > -- > Sincerely regards, > Ryanny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/eededd36/attachment-0002.html From john_platts at hotmail.com Tue Dec 1 12:46:02 2009 From: john_platts at hotmail.com (John Platts) Date: Tue, 1 Dec 2009 14:46:02 -0600 Subject: [Freeswitch-users] Blind transfer fails in FreeSWITCH, even if proxying and media bypass are enabled Message-ID: I have tried to do a blind transfer from a phone that is registered with FreeSWITCH, and it will fail, even when proxying and media bypass are enabled. Details about this issue can be found here: http://jira.freeswitch.org/browse/MODENDP-272 _________________________________________________________________ Get gifts for them and cashback for you. Try Bing now. http://www.bing.com/shopping/search?q=xbox+games&scope=cashback&form=MSHYCB&publ=WLHMTAG&crea=TEXT_MSHYCB_Shopping_Giftsforthem_cashback_1x1 From dome at tel.co.th Tue Dec 1 12:47:36 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Dec 2009 03:47:36 +0700 Subject: [Freeswitch-users] Cpu peak to 100% Openzap E1 R2 In-Reply-To: References: <8ccbff060912010543p5241831cg4c7f616e35b0294c@mail.gmail.com> <8ccbff060912010902t657f2905r4a4071fe92db8083@mail.gmail.com> Message-ID: <8ccbff060912011247j318ca089n6f0c9fa39c417dcd@mail.gmail.com> 2009/12/2 Moises Silva > > > On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote: > >> >> >> 2009/12/1 Moises Silva >> >> On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote: >>> >>>> Dear All, >>>> I got problem about openzap. I plan to use 4E1 R2 (Phone EQ >>>> card). I use zaptel driver from http://e400p.phoniceq.com/driver/(1.4.12.1) and freeswitch 1.0.5 pre 7 >>>> >>>> My Server Intel(R) Core(TM)2 Quad CPU Q6600 @ 2.40GHz >>>> and 2 GB memmory. >>>> when i start freeswitch if enable echo cencel in zt.conf CPU peak to >>>> 100% when i disable echo cancel FS can start but when first call incominng >>>> got 100% again. >>>> >>>> >>>> Someone can help me? >>>> >>>> >>>> >>> I suggest you to start using freeswitch trunk (and openzap trunk). >>> >>> >> I'll try >> >> >>> In the other hand, when you talk about 4E1 R2, you mean R2 as the >>> telephony signaling or is this some kind of board brand? >>> >> >> >> telephony signaling >> > > Does the signaling stack comes with the board and is integrated into > freeswitch as an endpoint? if not, your only chance is using openr2, which > requires freeswitch, openzap and openr2 trunk. > > http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 > yes i use openr2. I'm setup follow wiki. Dome C. > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0e241d08/attachment-0002.html From b0ef at esben-stien.name Tue Dec 1 14:27:49 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Tue, 01 Dec 2009 23:27:49 +0100 Subject: [Freeswitch-users] Freeswitch Video Capture and Playback In-Reply-To: <87k4xlga1k.fsf@quasar.esben-stien.name> (Esben Stien's message of "Fri\, 20 Nov 2009 04\:16\:55 +0100") References: <87k4xlga1k.fsf@quasar.esben-stien.name> Message-ID: <87pr6yz5wa.fsf@quasar.esben-stien.name> Esben Stien writes: > trying to record and play back video So nobody is using video with freeswitch?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From b0ef at esben-stien.name Tue Dec 1 14:30:09 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Tue, 01 Dec 2009 23:30:09 +0100 Subject: [Freeswitch-users] Ring Forever Message-ID: <87ljhmz5se.fsf@quasar.esben-stien.name> I'd like to set up an extension that would just ring forever. When a person calls this extension, it would ring until the end of times. I've tried several ways to do this, without luck, and I don't find any information on this on the wiki. Any pointers how to do this?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From erandr-junk at usa.net Tue Dec 1 13:51:42 2009 From: erandr-junk at usa.net (eaf) Date: Tue, 1 Dec 2009 13:51:42 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <26594250.post@talk.nabble.com> Hi, I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how it can be programmed), but ran into one issue with sound quality that I just cannot workaround by myself. I would describe the sound problem as being "choppy". From time to time small portions of the other party's voice are dropped, so the voice kind of stutters. This is not too bad, but is really noticeable, happens in every call and I don't experience the same with Asterisk running on the same box. I attached two files: freeswitch.wav and asterisk.mp3 to illustrate my point. Issue completely goes away, if I set inbound-proxy-media to true. The way how I test is to connect SPA-2000 via 10mbps LAN to the box directly exposed to internet, and then dial a toll-free via FutureNine (a SIP provider). The codec in use is PCMU. Can't really try PCMA or anything else with this provider. Only PCMU. Tried to match ptime of provider (30) with ptime of the SPA, didn't get any improvement. Tried turning off recording, no change either. What puzzles me is that even with greedy codec negotiations and with PCMU on both sides of FreeSWITCH, it's still saying that TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to illustrate. The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that it's not a performance issue. http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log Tried both 1.0.4 and 1.0.5pre5. Same results. What should I do next? Calls are consistently bad with FreeSWITCH, and consistently show no glitches with Asterisk. -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26594250.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From erandr-junk at usa.net Tue Dec 1 13:52:03 2009 From: erandr-junk at usa.net (eaf) Date: Tue, 1 Dec 2009 13:52:03 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <26599565.post@talk.nabble.com> I should also add, after browsing through some topics here, that my SIP provider sends 172-byte RTP frames, which is in accordance with ptime:20 that it gives to FreeSWITCH. eaf wrote: > > Hi, > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how > it can be programmed), but ran into one issue with sound quality that I > just cannot workaround by myself. I would describe the sound problem as > being "choppy". From time to time small portions of the other party's > voice are dropped, so the voice kind of stutters. This is not too bad, but > is really noticeable, happens in every call and I don't experience the > same with Asterisk running on the same box. I attached two files: > freeswitch.wav and asterisk.mp3 to illustrate my point. > > Issue completely goes away, if I set inbound-proxy-media to true. > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > directly exposed to internet, and then dial a toll-free via FutureNine (a > SIP provider). > > The codec in use is PCMU. Can't really try PCMA or anything else with this > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > the SPA, didn't get any improvement. Tried turning off recording, no > change either. > > What puzzles me is that even with greedy codec negotiations and with PCMU > on both sides of FreeSWITCH, it's still saying that > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log to > illustrate. > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > it's not a performance issue. > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > What should I do next? Calls are consistently bad with FreeSWITCH, and > consistently show no glitches with Asterisk. > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Tue Dec 1 14:05:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 16:05:45 -0600 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <87ljhmz5se.fsf@quasar.esben-stien.name> References: <87ljhmz5se.fsf@quasar.esben-stien.name> Message-ID: <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> do you want to generate ringback forever or no? The calling party will probably abort at some point. put both of these in your context then use one of these 2 sets of actions in your main ext On Tue, Dec 1, 2009 at 4:30 PM, Esben Stien wrote: > I'd like to set up an extension that would just ring forever. > > When a person calls this extension, it would ring until the end of > times. > > I've tried several ways to do this, without luck, and I don't find any > information on this on the wiki. > > Any pointers how to do this?. > > -- > Esben Stien is b0ef at e s a > http://www. s t n m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@ n n > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/3d01646d/attachment-0002.html From JCasale at activenetwerx.com Tue Dec 1 14:09:43 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 1 Dec 2009 22:09:43 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> Message-ID: >In this case, the $1 will only contain whatever is in the parens in your expression, i.e. > > >What do you have for your expression? >-MC Well, untested of course as I am busy with school:) But what I wrote up to try at Christmas (with your addition) was: Am I correct in presuming that Freeswitch will answer a fax from a local zap based user just like it does from an FXO port connected to a POTS line? What I hope to do here is catch any call made from that extension (the zap based fax machine/user) and push its call into the fax module. Thanks for taking the time to help! jlc From msc at freeswitch.org Tue Dec 1 14:22:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 14:22:53 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> Message-ID: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale wrote: > >In this case, the $1 will only contain whatever is in the parens in your > expression, i.e. > > > > > >What do you have for your expression? > >-MC > > Well, untested of course as I am busy with school:) But what I wrote up to > try at > Christmas (with your addition) was: > > > > > > > > data="/opt/freeswitch/scripts/emailfax.sh $1 at fax.com/tmp/${uuid}.rxfax.tiff"/> > > > > > Am I correct in presuming that Freeswitch will answer a fax from a local > zap based user > just like it does from an FXO port connected to a POTS line? What I hope to > do here is > catch any call made from that extension (the zap based fax machine/user) > and push its > call into the fax module. > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that dials. This extension looks ok. Try it out and let us know how it goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/927084aa/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 1 14:29:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 16:29:26 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26599565.post@talk.nabble.com> References: <26599565.post@talk.nabble.com> Message-ID: <191c3a030912011429p2cc7d834t9ca2b56b45682fe6@mail.gmail.com> linksys has had a bug for eons that can be fixed by setting the ptime (or rtp packet size in their terms) in it's firmware to .20 instead of .30 Asterisk does not use async RTP like we do so it's never a problem you can disable the timer by setting the channel var rtp_timer_name=none or sofia param rtp-timer-name to none in the sofia profile. You should also test this on latest SVN trunk or wait for pre8 On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > I should also add, after browsing through some topics here, that my SIP > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > that it gives to FreeSWITCH. > > > eaf wrote: > > > > Hi, > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > how > > it can be programmed), but ran into one issue with sound quality that I > > just cannot workaround by myself. I would describe the sound problem as > > being "choppy". From time to time small portions of the other party's > > voice are dropped, so the voice kind of stutters. This is not too bad, > but > > is really noticeable, happens in every call and I don't experience the > > same with Asterisk running on the same box. I attached two files: > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > directly exposed to internet, and then dial a toll-free via FutureNine (a > > SIP provider). > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > this > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > > the SPA, didn't get any improvement. Tried turning off recording, no > > change either. > > > > What puzzles me is that even with greedy codec negotiations and with PCMU > > on both sides of FreeSWITCH, it's still saying that > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > to > > illustrate. > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > it's not a performance issue. > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > consistently show no glitches with Asterisk. > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/09f705a3/attachment-0002.html From peter at cindyandpeter.com Tue Dec 1 14:41:38 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 17:41:38 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: <00ba01ca72d7$73272200$59756600$@com> Just remove the terminating '/' at the end of the second condition tag.... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 01, 2009 5:23 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Faxing Advice On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale wrote: >In this case, the $1 will only contain whatever is in the parens in your expression, i.e. > > >What do you have for your expression? >-MC Well, untested of course as I am busy with school:) But what I wrote up to try at Christmas (with your addition) was: Am I correct in presuming that Freeswitch will answer a fax from a local zap based user just like it does from an FXO port connected to a POTS line? What I hope to do here is catch any call made from that extension (the zap based fax machine/user) and push its call into the fax module. Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that dials. This extension looks ok. Try it out and let us know how it goes. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1131e3e6/attachment-0002.html From JCasale at activenetwerx.com Tue Dec 1 15:08:06 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 1 Dec 2009 23:08:06 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <00ba01ca72d7$73272200$59756600$@com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> <00ba01ca72d7$73272200$59756600$@com> Message-ID: >Just remove the terminating '/' at the end of the second condition tag.... > > I tried to see based on examples if it was obvious to me why that should not be there but it didn't jump out:) Cuold you explain that please? The gateway will arrive at the end of the week, but I probably won't get to this now until Christmas as I missed my opportunity:) Once I get it working, I will update the wiki as I am sure it's useful to many others. Thanks everyone! jlc From Russell.Mosemann at cune.org Tue Dec 1 15:21:33 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 1 Dec 2009 23:21:33 -0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: Message-ID: <20091201232133.76DAB4309FA@mail.cune.org> "Joseph L. Casale" said: > >Just remove the terminating '/' at the end of the second condition tag.... > > > > > > I tried to see based on examples if it was obvious to me why that should not be there > but it didn't jump out:) Cuold you explain that please? It is a multi-line condition. If a condition is only one line long, it begins and ends on the same line. The "/>" combination is the sequence that closes the tag. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From b0ef at esben-stien.name Tue Dec 1 16:29:25 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Wed, 02 Dec 2009 01:29:25 +0100 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> (Anthony Minessale's message of "Tue\, 1 Dec 2009 16\:05\:45 -0600") References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> Message-ID: <87einexlp6.fsf@quasar.esben-stien.name> Anthony Minessale writes: > do you want to generate ringback forever or no? Yes, I want the dialing party to get a ring forever. This is because I cannot transfer the party to my SIP phone, because my SIP phone is broken for incoming calls. I'll solve it by letting the party hear a ring and then I'll use the event socket to give me a notification. I'll then dial in with my SIP phone to a conference room and transfer the calling party there. I did this: ..which gives me an eternal ring, but it sounds choppy and it sounds as if the following ring starts on top of the currently playing ring. I hear crackling sounds, which I don't experience with any other use of freeswitch. Any idea? -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From anthony.minessale at gmail.com Tue Dec 1 15:39:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 17:39:14 -0600 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <87einexlp6.fsf@quasar.esben-stien.name> References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> <87einexlp6.fsf@quasar.esben-stien.name> Message-ID: <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> are you on older REV? try answering first to compare. On Tue, Dec 1, 2009 at 6:29 PM, Esben Stien wrote: > Anthony Minessale writes: > > > do you want to generate ringback forever or no? > > Yes, I want the dialing party to get a ring forever. This is because I > cannot transfer the party to my SIP phone, because my SIP phone is > broken for incoming calls. > > I'll solve it by letting the party hear a ring and then I'll use the > event socket to give me a notification. I'll then dial in with my SIP > phone to a conference room and transfer the calling party there. > > I did this: > > > > > > > > > > > > > > > > ..which gives me an eternal ring, but it sounds choppy and it sounds as > if the following ring starts on top of the currently playing ring. > > I hear crackling sounds, which I don't experience with any other use of > freeswitch. > > Any idea? > > -- > Esben Stien is b0ef at e s a > http://www. s t n m > irc://irc. b - i . e/%23contact > sip:b0ef@ e e > jid:b0ef@ n n > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/ce79f411/attachment-0002.html From b0ef at esben-stien.name Tue Dec 1 17:24:11 2009 From: b0ef at esben-stien.name (Esben Stien) Date: Wed, 02 Dec 2009 02:24:11 +0100 Subject: [Freeswitch-users] Ring Forever In-Reply-To: <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> (Anthony Minessale's message of "Tue\, 1 Dec 2009 17\:39\:14 -0600") References: <87ljhmz5se.fsf@quasar.esben-stien.name> <191c3a030912011405j57f3a8a3k5b5bbda41957dd8b@mail.gmail.com> <87einexlp6.fsf@quasar.esben-stien.name> <191c3a030912011539k3d5af90fy967bf6ef55820704@mail.gmail.com> Message-ID: <87einew4lg.fsf@quasar.esben-stien.name> Anthony Minessale writes: > are you on older REV? I think I'm on 15334 > try answering first to compare. ?, that's what I do: -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n From msc at freeswitch.org Tue Dec 1 16:56:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Dec 2009 16:56:25 -0800 Subject: [Freeswitch-users] FreeSWITCH Survey: What Environment Do You Normally Use? Message-ID: <87f2f3b90912011656m5d08d466n641e341ec9889373@mail.gmail.com> Hi folks, I'm doing a little survey to get an idea of what everyone prefers to use for their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc. Please log in to the main page and check out this node: http://www.freeswitch.org/node/206 Select the environment that you use the most or prefer to use. In a week or so I will send out the final tally. I'm sure the only question is who will come in second place after 64-bit CentOS/Red Hat. :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/1d71d6b5/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 1 17:02:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Dec 2009 19:02:11 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: References: Message-ID: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: > Sure, I'll try that. I'm just building freeswitch-snapshot that I > downloaded from files.freeswitch.org > > I also experience, when bridging a call from an FXS to FXO the call is cut > after a random time (this does not appear when bridging SIP to FXO). Might > this upgrade fix this problem also ? > > > > Fran?ois > > > > On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: > > can you test svn trunk or latest pre release of 1.0.5 > > > On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: > >> Hello, >> >> >> >> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP >> problems on the FXS ports. >> >> When I ring on FXS ports, the connected phone does not display >> callerid/callerid-name. >> >> I tried turning the stuff of in openzap.conf.xml () but it did not help. >> >> >> >> As a side note, turning this on on the FXO ports drops the callerid >> information on incoming calls. >> >> >> >> Running freeswitch 1.0.4 on linux 2.6.27. >> >> >> >> Fran?ois >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/edb26ac0/attachment-0002.html From erandr-junk at usa.net Tue Dec 1 17:26:58 2009 From: erandr-junk at usa.net (erandr-junk at usa.net) Date: Tue, 01 Dec 2009 20:26:58 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <927NLBBZ73918S04.1259717218@cmsweb04.cms.usa.net> Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just installing SVN trunk didn't fix it either, but setting that option afterwards surely did the trick. One thing I've noticed while staring at the console is that it *looks like* that w/o the new setting the stuttering happens when FS either re-registers itself with the provider or one of the SPA's port re-registers with FS. ------ Original Message ------ Received: Tue, 01 Dec 2009 05:33:26 PM EST From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Choppy sound with PCMU > linksys has had a bug for eons that can be fixed by setting the ptime (or > rtp packet size in their terms) > in it's firmware to .20 instead of .30 > > Asterisk does not use async RTP like we do so it's never a problem > you can disable the timer by setting the channel var rtp_timer_name=none or > sofia param rtp-timer-name to none in the sofia profile. > > You should also test this on latest SVN trunk or wait for pre8 > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > I should also add, after browsing through some topics here, that my SIP > > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > > that it gives to FreeSWITCH. > > > > > > eaf wrote: > > > > > > Hi, > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > > how > > > it can be programmed), but ran into one issue with sound quality that I > > > just cannot workaround by myself. I would describe the sound problem as > > > being "choppy". From time to time small portions of the other party's > > > voice are dropped, so the voice kind of stutters. This is not too bad, > > but > > > is really noticeable, happens in every call and I don't experience the > > > same with Asterisk running on the same box. I attached two files: > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > directly exposed to internet, and then dial a toll-free via FutureNine (a > > > SIP provider). > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > > this > > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime of > > > the SPA, didn't get any improvement. Tried turning off recording, no > > > change either. > > > > > > What puzzles me is that even with greedy codec negotiations and with PCMU > > > on both sides of FreeSWITCH, it's still saying that > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > > to > > > illustrate. > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800 > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > > it's not a performance issue. > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > > consistently show no glitches with Asterisk. > > > > > > > > > > -- > > View this message in context: > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peter at cindyandpeter.com Tue Dec 1 18:26:26 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 21:26:26 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <20091201232133.76DAB4309FA@mail.cune.org> References: <20091201232133.76DAB4309FA@mail.cune.org> Message-ID: <00dd01ca72f6$da19ddd0$8e4d9970$@com> To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. ... content ... If there is no content, you get: Or, on one line, . You're allowed to abbreviate that to just . So in your case: <--- these are themselves abbreviations of !! Hope that helps! Peter -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Russell.Mosemann at cune.org Sent: Tuesday, December 01, 2009 6:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Faxing Advice "Joseph L. Casale" said: > >Just remove the terminating '/' at the end of the second condition tag.... > > > > > > I tried to see based on examples if it was obvious to me why that should not be there > but it didn't jump out:) Cuold you explain that please? It is a multi-line condition. If a condition is only one line long, it begins and ends on the same line. The "/>" combination is the sequence that closes the tag. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From JCasale at activenetwerx.com Tue Dec 1 19:02:13 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 2 Dec 2009 03:02:13 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <00dd01ca72f6$da19ddd0$8e4d9970$@com> References: <20091201232133.76DAB4309FA@mail.cune.org> <00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: >To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. > > ... content ... > /snip Ahh, so must all the actions be contained within at least one condition tag as content, or could have I kept the last "/" on the last condition and dropped the line? Thanks everyone :) jlc From juanbackson at gmail.com Tue Dec 1 19:11:37 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Dec 2009 11:11:37 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> Message-ID: <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Hi, I also did try to set only bypass_media, but it still does not work? freeswitch still modifies the c= line, causing the call to fail. Could someone please help? send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a Max-Forwards: 69 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/ 90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [calling][0] 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/ 90964111 at 192.168.1.13:5060 SOFIA ROUTING 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [proceeding][180] 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/external_1/90964111 at 192.168.1.13:5060! 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/external_1/PhonerLite at 192.168.1.102! 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 From: >;tag=2454193703 To: >;tag=FFKXgjN02m02N Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [early][180] 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a From: PhonerLite >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 INVITE Contact: Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 256 v=0 o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10096 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp Max-Forwards: 70 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757372 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [ready][200] 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10096 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/ 90964111 at 192.168.1.13:5060] has been answered 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/ 90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 From: >;tag=2454193703 To: >;tag=FFKXgjN02m02N2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 originate_disposition=[SUCCESS] Content-Type: application/sdp Content-Disposition: session 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] Content-Length: 207 v=0 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] m=audio 10096 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/ 90964111 at 192.168.1.13:5060 SOFIA HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next route... EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_HIBERNATE] Channel-State-Number: [8] Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [answered] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259751417450336] Caller-Channel-Created-Time: [1259751417450336] Caller-Channel-Answered-Time: [1259751419381828] Caller-Channel-Progress-Time: [1259751417736234] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2454193703] variable_sofia_profile_name: [external_1] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 794697697 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ >;tag=2454193703 To: >;tag=FFKXgjN02m02N Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] Contact: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/ 90964111 at 192.168.1.116:9392) 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/ PhonerLite at 192.168.1.102 receive message [MEDIA] 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/ PhonerLite at 192.168.1.102 reading on a session with no media! 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [completed][200] 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ PhonerLite at 192.168.1.102 entering state [ready][200] 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/ 90964111 at 192.168.1.13:5060 receive message [MEDIA] 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj Max-Forwards: 69 From: "PhonerLite" >;tag=jaZ7N37atF3tr To: ;tag=8849584 Call-ID: 40563247-59d4-122d-ff84-0022190e9476 CSeq: 123757373 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off v=0 o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 30632 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/ 90964111 at 192.168.1.13:5060 entering state [calling][0] send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: ------------------------------------------------------------------------ On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > yes he did you can see it in his trace. > you can not use both of them together...... > > > > On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > >> The only way this would happen would be if this is set to proxy media not >> bypass. Are you setting both? >> >> Mike >> >> On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: >> >> In the following trace, 102 is FS IP, 104 is calling party and 13 is >> called party. >> >> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >> >> Any idea why? >> >> >> freeswitch at localhost.localdomain> recv 951 bytes from >> udp/[192.168.1.104]:5060 at 22:56:33.782715: >> >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.104:5060 >> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >> From: >> >;tag=786224322 >> To: > >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> Contact: >> Content-Type: application/sdp >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> Max-Forwards: 70 >> Supported: 100rel, replaces >> User-Agent: SIPPER for PhonerLite >> Content-Length: 397 >> >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.104:5060 >> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >> From: >> >;tag=786224322 >> To: > >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel >> sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ >> PhonerLite at 192.168.1.102 entering state [received][100] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> >> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_EXECUTE] >> Channel-State-Number: [4] >> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [ringing] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259708193783162] >> Caller-Channel-Created-Time: [1259708193783162] >> Caller-Channel-Answered-Time: [0] >> Caller-Channel-Progress-Time: [0] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [786224322] >> variable_sofia_profile_name: [internal] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> ] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_bypass_media: [true] >> variable_proxy_media: [true] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> >> >> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ >> 90964111 at 192.168.1.116:9390 Patched SDP >> --- >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> +++ >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >> (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to >> sleep >> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >> >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >> Max-Forwards: 69 >> From: "PhonerLite" >> >;tag=8tH6Xjt2XaU9F >> To: >> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >> CSeq: 123735760 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 404 >> Remote-Party-ID: "PhonerLite" >> >;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/963a7360/attachment-0002.html From erandr-junk at usa.net Tue Dec 1 19:19:39 2009 From: erandr-junk at usa.net (erandr-junk at usa.net) Date: Tue, 01 Dec 2009 22:19:39 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU Message-ID: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> Wow... Thinking about this timer setting and about how it converted send()/recv() from non-blocking to blocking, I straced freeswitch when it was supposed to be idle. It never pauses! It keeps going in and out of select() every millisecond! Why?? ------ Original Message ------ Received: Tue, 01 Dec 2009 08:31:46 PM EST From: erandr-junk at usa.net To: Subject: Re: [Freeswitch-users] Choppy sound with PCMU > Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just > installing SVN trunk didn't fix it either, but setting that option afterwards > surely did the trick. > > One thing I've noticed while staring at the console is that it *looks like* > that w/o the new setting the stuttering happens when FS either re-registers > itself with the provider or one of the SPA's port re-registers with FS. > > ------ Original Message ------ > Received: Tue, 01 Dec 2009 05:33:26 PM EST > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > linksys has had a bug for eons that can be fixed by setting the ptime (or > > rtp packet size in their terms) > > in it's firmware to .20 instead of .30 > > > > Asterisk does not use async RTP like we do so it's never a problem > > you can disable the timer by setting the channel var rtp_timer_name=none or > > sofia param rtp-timer-name to none in the sofia profile. > > > > You should also test this on latest SVN trunk or wait for pre8 > > > > > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > > > > I should also add, after browsing through some topics here, that my SIP > > > provider sends 172-byte RTP frames, which is in accordance with ptime:20 > > > that it gives to FreeSWITCH. > > > > > > > > > eaf wrote: > > > > > > > > Hi, > > > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the way > > > how > > > > it can be programmed), but ran into one issue with sound quality that I > > > > just cannot workaround by myself. I would describe the sound problem as > > > > being "choppy". From time to time small portions of the other party's > > > > voice are dropped, so the voice kind of stutters. This is not too bad, > > > but > > > > is really noticeable, happens in every call and I don't experience the > > > > same with Asterisk running on the same box. I attached two files: > > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > > directly exposed to internet, and then dial a toll-free via FutureNine > (a > > > > SIP provider). > > > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else with > > > this > > > > provider. Only PCMU. Tried to match ptime of provider (30) with ptime > of > > > > the SPA, didn't get any improvement. Tried turning off recording, no > > > > change either. > > > > > > > > What puzzles me is that even with greedy codec negotiations and with > PCMU > > > > on both sides of FreeSWITCH, it's still saying that > > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of freeswitch.log > > > to > > > > illustrate. > > > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode > LX800 > > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that > > > > it's not a performance issue. > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav > > > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log > > > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, and > > > > consistently show no glitches with Asterisk. > > > > > > > > > > > > > > -- > > > View this message in context: > > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Dec 1 19:26:27 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 22:26:27 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Message-ID: how are you sending both invites here? can you explain the full call path and how you are originating these calls? On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > Hi, > > I also did try to set?only bypass_media, but it still does not work??freeswitch still modifies the c= line, causing the call to fail. > > Could someone please help? > > > send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > Max-Forwards: 69 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 402 > Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/90964111 at 192.168.1.13:5060 SOFIA ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep > recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [proceeding][180] > 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/external_1/90964111 at 192.168.1.13:5060! > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/external_1/PhonerLite at 192.168.1.102! > 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [early][180] > 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! > recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Supported: replaces > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: > ------------------------------------------------------------------------ > ACK sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp > Max-Forwards: 70 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [ready][200] > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/90964111 at 192.168.1.13:5060] has been answered > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) > EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) > > 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] > send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE > > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 originate_disposition=[SUCCESS] > Content-Type: application/sdp > Content-Disposition: session > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > Content-Length: 207 > > v=0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > m=audio 10096 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() > 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next route... > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() > 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_HIBERNATE] > Channel-State-Number: [8] > Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [answered] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259751417450336] > Caller-Channel-Created-Time: [1259751417450336] > Caller-Channel-Answered-Time: [1259751419381828] > Caller-Channel-Progress-Time: [1259751417736234] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > Other-Leg-Username: [PhonerLite] > Other-Leg-Caller-ID-Name: [PhonerLite] > Other-Leg-Caller-ID-Number: [PhonerLite] > Other-Leg-Network-Addr: [192.168.1.104] > Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] > Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] > Other-Leg-Source: [mod_sofia] > Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] > Other-Leg-Screen-Bit: [true] > Other-Leg-Privacy-Hide-Name: [false] > Other-Leg-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [2454193703] > variable_sofia_profile_name: [external_1] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] > variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 794697697 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > variable_proto_specific_hangup_cause: [sip:403] > variable_sip_hangup_phrase: [Because] > variable_bypass_media: [true] > variable_success_bridge: [true] > variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] > variable_switch_m_sdp: [v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_endpoint_disposition: [ANSWER] > variable_originate_disposition: [SUCCESS] > variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] > variable_current_application: [info] > > recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: > ------------------------------------------------------------------------ > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport > From: ;tag=2454193703 > To: ;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] > > Contact: > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/90964111 at 192.168.1.116:9392) > 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/PhonerLite at 192.168.1.102 receive message [MEDIA] > 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/PhonerLite at 192.168.1.102 reading on a session with no media! > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [completed][200] > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/PhonerLite at 192.168.1.102 entering state [ready][200] > 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/90964111 at 192.168.1.13:5060 receive message [MEDIA] > 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: > ------------------------------------------------------------------------ > INVITE sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj > Max-Forwards: 69 > From: "PhonerLite" ;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757373 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 223 > Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 30632 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: > ------------------------------------------------------------------------ > > On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale wrote: > yes he did you can see it in his trace. > you can not use both of them together...... > > > > On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: > The only way this would happen would be if this is set to proxy media not bypass. Are you setting both? > > Mike > > On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: > >> In the following trace, 102 is FS IP, 104 is calling party and 13 is called party. >> >> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >> >> Any idea why? >> >> >> freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 22:56:33.782715: >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >> From: ;tag=786224322 >> To: >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> Contact: >> Content-Type: application/sdp >> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE >> Max-Forwards: 70 >> Supported: 100rel, replaces >> User-Agent: SIPPER for PhonerLite >> Content-Length: 397 >> >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=sendrecv >> ------------------------------------------------------------------------ >> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >> From: ;tag=786224322 >> To: >> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >> CSeq: 37 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Content-Length: 0 >> >> ------------------------------------------------------------------------ >> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [received][100] >> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> >> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_EXECUTE] >> Channel-State-Number: [4] >> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [ringing] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259708193783162] >> Caller-Channel-Created-Time: [1259708193783162] >> Caller-Channel-Answered-Time: [0] >> Caller-Channel-Progress-Time: [0] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [786224322] >> variable_sofia_profile_name: [internal] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_bypass_media: [true] >> variable_proxy_media: [true] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> >> >> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/90964111 at 192.168.1.116:9390 Patched SDP >> --- >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> +++ >> v=0 >> o=- 3393406017 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA ROUTING >> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep >> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >> ------------------------------------------------------------------------ >> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >> Max-Forwards: 69 >> From: "PhonerLite" ;tag=8tH6Xjt2XaU9F >> To: >> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >> CSeq: 123735760 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 404 >> Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/a6b9d30c/attachment-0002.html From Russell.Mosemann at cune.org Tue Dec 1 19:34:03 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Tue, 1 Dec 2009 21:34:03 -0600 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <20091201232133.76DAB4309FA@mail.cune.org><00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: Joseph L. Casale wrote: > Ahh, so must all the actions be contained within at least one condition > tag as content, Yes. > or could have I kept the > last "/" on the last condition and dropped the line? No. Think of the tags as a begin/end pair that surround the content. If there is no content, then you can use a one-line condition tag. or stuff -- Russell Mosemann From peter at cindyandpeter.com Tue Dec 1 19:41:32 2009 From: peter at cindyandpeter.com (Peter J. Zandvoort) Date: Tue, 1 Dec 2009 22:41:32 -0500 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <20091201232133.76DAB4309FA@mail.cune.org> <00dd01ca72f6$da19ddd0$8e4d9970$@com> Message-ID: <00e401ca7301$5723b580$056b2080$@com> Indeed, all actions must be contained with a condition tag. FS just processes the conditions inside the extension top-to-bottom, so: if the top condition (without actions in it) doesn't match, it stops processing that extension. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph L. Casale Sent: Tuesday, December 01, 2009 10:02 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: [Freeswitch-users] Faxing Advice >To expand on what Russell said: XML always has a start and an end tag, possibly with other stuff in between. > > ... content ... > /snip Ahh, so must all the actions be contained within at least one condition tag as content, or could have I kept the last "/" on the last condition and dropped the line? Thanks everyone :) jlc _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From juanbackson at gmail.com Tue Dec 1 20:07:22 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Dec 2009 12:07:22 +0800 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> Message-ID: <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> Hi Mike, Here is a very strange SIP outgoing INVITES from freeswitch: The call path is this: 192.168.1.104 (phonerlite) -> 192.168.1.102 ( freeswitch ) 192.168.1.102 -> 192.168.1.116 (sipp gives back 403) 192.168.1.102 -> 192.168.1.13 ( phone ) The first INVITE to 192.168.1.13 has the right c= and o= ( both is pointing to 192.168.1.104). But the for some unknown reason, Freeswitch sends INVITES again. But in the INVITE resend, the o = becomes fs's ip. I have no idea. This is only bypass_meida. Attached is the fs log. ngrep -q -p -W byline port 5060 interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip) and ( port 5060 ) U 192.168.1.104:5060 -> 192.168.1.102:5060 INVITE sip:90964111 at 192.168.1.102 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport. From: >;tag=2563216860. To: >. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . Content-Type: application/sdp. Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE. Max-Forwards: 70. Supported: 100rel, replaces. User-Agent: SIPPER for PhonerLite. Content-Length: 396. . v=0. o=- 478760567 0 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=sendrecv. U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9390 INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. Max-Forwards: 69. From: "PhonerLite" >;tag=r0pv05c0848ae. To: . Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 401. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 523740055483911509 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.116:9390 -> 192.168.1.102:5060 SIP/2.0 403 Because. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. From: "PhonerLite" >;tag=r0pv05c0848ae. To: ;tag=6. Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9390 ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. Max-Forwards: 69. From: "PhonerLite" >;tag=r0pv05c0848ae. To: ;tag=6. Call-ID: a76ed230-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9391 INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. Max-Forwards: 69. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: . Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 402. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 5392773558290384508 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.116:9391 -> 192.168.1.102:5060 SIP/2.0 403 Because. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: ;tag=6. Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.116:9391 ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. Max-Forwards: 69. From: "PhonerLite" >;tag=S9FN20X35DZXS. To: ;tag=6. Call-ID: a771eccc-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: . Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 402. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522444 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.104. t=0 0. m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=rtpmap:110 speex/8000. a=rtpmap:111 speex/16000. a=rtpmap:9 G722/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: . Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 INVITE. Contact: . Supported: replaces. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE. Content-Type: application/sdp. Content-Length: 256. . v=0. o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. s=A conversation. c=IN IP4 192.168.1.13. t=0 0. m=audio 10098 RTP/AVP 8 0 9 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 192.168.1.102:5060 -> 192.168.1.13:5060 ACK sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ. Max-Forwards: 70. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758962 ACK. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 207. . v=0. o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. s=A conversation. c=IN IP4 192.168.1.13. t=0 0. m=audio 10098 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 192.168.1.104:5060 -> 192.168.1.102:5060 ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104:5060 ;branch=z9hG4bK00b67e2664ddde1180d3001a805656a5;rport. From: >;tag=2563216860. To: >;tag=QQX3yavvBvjrj. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 17 ACK. Contact: . Max-Forwards: 70. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 223. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.102. t=0 0. m=audio 33352 RTP/AVP 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 192.168.1.104:43488 -> 192.168.1.102:5060 . . .............. U 192.168.1.102:5060 -> 192.168.1.13:5060 INVITE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 223. Remote-Party-ID: "PhonerLite" >;party=calling;screen=yes;privacy=off. . v=0. o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. s=SIPPER for PhonerLite. c=IN IP4 192.168.1.102. t=0 0. m=audio 33352 RTP/AVP 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 ACK sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. Max-Forwards: 69. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 ACK. Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.13:5060 BYE sip:192.168.1.13:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e. Max-Forwards: 70. From: "PhonerLite" >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758964 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". Content-Length: 0. . U 192.168.1.102:5060 -> 192.168.1.104:5060 BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma. Max-Forwards: 70. From: >;tag=QQX3yavvBvjrj. To: >;tag=2563216860. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 123758964 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". Content-Length: 0. . U 192.168.1.104:5060 -> 192.168.1.102:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma. From: >;tag=QQX3yavvBvjrj. To: >;tag=2563216860. Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. CSeq: 123758964 BYE. Contact: . User-Agent: SIPPER for PhonerLite. Content-Length: 0. . U 192.168.1.13:5060 -> 192.168.1.102:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. From: PhonerLite >;tag=tj9D4Ue72pNgN. To: ;tag=2900712955. Call-ID: a776ce22-59db-122d-ff84-0022190e9476. CSeq: 123758963 INVITE. Contact: . Content-Length: 0. . On Wed, Dec 2, 2009 at 11:26 AM, Michael Jerris wrote: > how are you sending both invites here? can you explain the full call path > and how you are originating these calls? > > > On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > > Hi, > > I also did try to set only bypass_media, but it still does not work? > freeswitch still modifies the c= line, causing the call to fail. > > Could someone please help? > > > send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: > ------------------------------------------------------------------------ > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > Max-Forwards: 69 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 402 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/external_1/ > 90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [calling][0] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 > (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 > (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/external_1/ > 90964111 at 192.168.1.13:5060 SOFIA ROUTING > 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 > (sofia/external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 > (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going to sleep > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 > (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA > 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 > (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to > sleep > recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [proceeding][180] > 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready > sofia/external_1/90964111 at 192.168.1.13:5060! > 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ > PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready > sofia/external_1/PhonerLite at 192.168.1.102! > 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: > >;tag=2454193703 > To: > >;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 > sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] > 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [early][180] > 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 Ring Ready > sofia/external_1/PhonerLite at 192.168.1.102! > recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a > From: PhonerLite > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 INVITE > Contact: > Supported: replaces > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, > UPDATE, MESSAGE > Content-Type: application/sdp > Content-Length: 256 > > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: > ------------------------------------------------------------------------ > ACK sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp > Max-Forwards: 70 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757372 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [ready][200] > 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: > v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/external_1/ > 90964111 at 192.168.1.13:5060] has been answered > 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/external_1/ > 90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 > 256 25 2591585 1) > EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 > 621 201 256 25 2591585 1) > > 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 Channel > [sofia/external_1/PhonerLite at 192.168.1.102] has been answered > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 Originate > Resulted in Success: [sofia/external_1/90964111 at 192.168.1.13:5060] > send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 > From: > >;tag=2454193703 > To: >;tag=FFKXgjN02m02N2009-12-02 > 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ > PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE > > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media=[true] > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 > originate_disposition=[SUCCESS] > Content-Type: application/sdp > Content-Disposition: session > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > Content-Length: 207 > > v=0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 > (sofia/external_1/90964111 at 192.168.1.13:5060) State Change > CS_CONSUME_MEDIA -> CS_HIBERNATE > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > m=audio 10096 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 > (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change > CS_HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 > (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/external_1/ > 90964111 at 192.168.1.13:5060 SOFIA HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 > sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE > 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 > (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going to > sleep > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() > 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to get next > route... > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() > 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: > Channel-State: [CS_HIBERNATE] > Channel-State-Number: [8] > Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Call-Direction: [inbound] > Presence-Call-Direction: [inbound] > Answer-State: [answered] > Caller-Username: [PhonerLite] > Caller-Dialplan: [class4] > Caller-Caller-ID-Name: [PhonerLite] > Caller-Caller-ID-Number: [PhonerLite] > Caller-Network-Addr: [192.168.1.104] > Caller-Destination-Number: [90964111] > Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] > Caller-Source: [mod_sofia] > Caller-Context: [default] > Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] > Caller-Profile-Index: [1] > Caller-Profile-Created-Time: [1259751417450336] > Caller-Channel-Created-Time: [1259751417450336] > Caller-Channel-Answered-Time: [1259751419381828] > Caller-Channel-Progress-Time: [1259751417736234] > Caller-Channel-Progress-Media-Time: [0] > Caller-Channel-Hangup-Time: [0] > Caller-Channel-Transfer-Time: [0] > Caller-Screen-Bit: [true] > Caller-Privacy-Hide-Name: [false] > Caller-Privacy-Hide-Number: [false] > Other-Leg-Username: [PhonerLite] > Other-Leg-Caller-ID-Name: [PhonerLite] > Other-Leg-Caller-ID-Number: [PhonerLite] > Other-Leg-Network-Addr: [192.168.1.104] > Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] > Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] > Other-Leg-Source: [mod_sofia] > Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] > Other-Leg-Screen-Bit: [true] > Other-Leg-Privacy-Hide-Name: [false] > Other-Leg-Privacy-Hide-Number: [false] > variable_sip_received_ip: [192.168.1.104] > variable_sip_received_port: [5060] > variable_sip_via_protocol: [udp] > variable_sip_from_user: [PhonerLite] > variable_sip_from_uri: [PhonerLite at 192.168.1.102] > variable_sip_from_host: [192.168.1.102] > variable_sip_from_user_stripped: [PhonerLite] > variable_sip_from_tag: [2454193703] > variable_sofia_profile_name: [external_1] > variable_sip_req_user: [90964111] > variable_sip_req_uri: [90964111 at 192.168.1.102] > variable_sip_req_host: [192.168.1.102] > variable_sip_to_user: [90964111] > variable_sip_to_uri: [90964111 at 192.168.1.102] > variable_sip_to_host: [192.168.1.102] > variable_sip_contact_port: [5060] > variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] > variable_sip_contact_host: [192.168.1.104] > variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] > variable_sip_call_id: [000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104] > variable_sip_user_agent: [SIPPER for PhonerLite] > variable_sip_via_host: [192.168.1.104] > variable_sip_via_port: [5060] > variable_sip_via_rport: [5060] > variable_max_forwards: [70] > variable_switch_r_sdp: [v=0 > o=- 794697697 0 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.104 > t=0 0 > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:111 speex/16000 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > ] > variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] > variable_effective_caller_id_number: [PhonerLite] > variable_effective_caller_id_name: [PhonerLite] > variable_ variable_routing_digit: [90964111] > variable_continue_on_fail: [true] > variable_hangup_after_bridge: [true] > variable_sip_contact_user: [PhonerLite] > variable_proto_specific_hangup_cause: [sip:403] > variable_sip_hangup_phrase: [Because] > variable_bypass_media: [true] > variable_success_bridge: [true] > variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] > variable_switch_m_sdp: [v=0 > o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 > s=A conversation > c=IN IP4 192.168.1.13 > t=0 0 > m=audio 10096 RTP/AVP 8 0 9 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:9 G722/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > ] > variable_endpoint_disposition: [ANSWER] > variable_originate_disposition: [SUCCESS] > variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] > variable_current_application: [info] > > recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: > ------------------------------------------------------------------------ > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.104:5060 > ;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport > From: > >;tag=2454193703 > To: > >;tag=FFKXgjN02m02N > Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 > CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 > sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]=[90964111] > > Contact: > Max-Forwards: 70 > Content-Length: 0 > > ------------------------------------------------------------------------ > > EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/external_1/ > 90964111 at 192.168.1.116:9392) > 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/external_1/ > PhonerLite at 192.168.1.102 receive message [MEDIA] > 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] > 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/external_1/ > PhonerLite at 192.168.1.102 reading on a session with no media! > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [completed][200] > 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > PhonerLite at 192.168.1.102 entering state [ready][200] > 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/external_1/ > 90964111 at 192.168.1.13:5060 receive message [MEDIA] > 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send signal > sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: > ------------------------------------------------------------------------ > INVITE sip:192.168.1.13:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj > Max-Forwards: 69 > From: "PhonerLite" > >;tag=jaZ7N37atF3tr > To: ;tag=8849584 > Call-ID: 40563247-59d4-122d-ff84-0022190e9476 > CSeq: 123757373 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 223 > Remote-Party-ID: "PhonerLite" > >;party=calling;screen=yes;privacy=off > > v=0 > o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 > s=SIPPER for PhonerLite > c=IN IP4 192.168.1.102 > t=0 0 > m=audio 30632 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ------------------------------------------------------------------------ > 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/external_1/ > 90964111 at 192.168.1.13:5060 entering state [calling][0] > send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: > ------------------------------------------------------------------------ > > On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> yes he did you can see it in his trace. >> you can not use both of them together...... >> >> >> >> On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote: >> >>> The only way this would happen would be if this is set to proxy media not >>> bypass. Are you setting both? >>> >>> Mike >>> >>> On Dec 1, 2009, at 10:08 AM, Juan Backson wrote: >>> >>> In the following trace, 102 is FS IP, 104 is calling party and 13 is >>> called party. >>> >>> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >>> >>> Any idea why? >>> >>> >>> freeswitch at localhost.localdomain> recv 951 bytes from >>> udp/[192.168.1.104]:5060 at 22:56:33.782715: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.104:5060 >>> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >>> From: >>> >;tag=786224322 >>> To: > >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> Contact: >>> Content-Type: application/sdp >>> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, >>> UPDATE >>> Max-Forwards: 70 >>> Supported: 100rel, replaces >>> User-Agent: SIPPER for PhonerLite >>> Content-Length: 397 >>> >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=sendrecv >>> >>> ------------------------------------------------------------------------ >>> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.1.104:5060 >>> ;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >>> From: >>> >;tag=786224322 >>> To: > >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New Channel >>> sofia/internal/PhonerLite at 192.168.1.102[d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/internal/ >>> PhonerLite at 192.168.1.102 entering state [received][100] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> >>> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >>> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >>> Channel-State: [CS_EXECUTE] >>> Channel-State-Number: [4] >>> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Call-Direction: [inbound] >>> Presence-Call-Direction: [inbound] >>> Answer-State: [ringing] >>> Caller-Username: [PhonerLite] >>> Caller-Dialplan: [class4] >>> Caller-Caller-ID-Name: [PhonerLite] >>> Caller-Caller-ID-Number: [PhonerLite] >>> Caller-Network-Addr: [192.168.1.104] >>> Caller-Destination-Number: [90964111] >>> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Caller-Source: [mod_sofia] >>> Caller-Context: [default] >>> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Caller-Profile-Index: [1] >>> Caller-Profile-Created-Time: [1259708193783162] >>> Caller-Channel-Created-Time: [1259708193783162] >>> Caller-Channel-Answered-Time: [0] >>> Caller-Channel-Progress-Time: [0] >>> Caller-Channel-Progress-Media-Time: [0] >>> Caller-Channel-Hangup-Time: [0] >>> Caller-Channel-Transfer-Time: [0] >>> Caller-Screen-Bit: [true] >>> Caller-Privacy-Hide-Name: [false] >>> Caller-Privacy-Hide-Number: [false] >>> variable_sip_received_ip: [192.168.1.104] >>> variable_sip_received_port: [5060] >>> variable_sip_via_protocol: [udp] >>> variable_sip_from_user: [PhonerLite] >>> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >>> variable_sip_from_host: [192.168.1.102] >>> variable_sip_from_user_stripped: [PhonerLite] >>> variable_sip_from_tag: [786224322] >>> variable_sofia_profile_name: [internal] >>> variable_sip_req_user: [90964111] >>> variable_sip_req_uri: [90964111 at 192.168.1.102] >>> variable_sip_req_host: [192.168.1.102] >>> variable_sip_to_user: [90964111] >>> variable_sip_to_uri: [90964111 at 192.168.1.102] >>> variable_sip_to_host: [192.168.1.102] >>> variable_sip_contact_port: [5060] >>> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >>> variable_sip_contact_host: [192.168.1.104] >>> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >>> variable_sip_call_id: [ >>> 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104] >>> variable_sip_user_agent: [SIPPER for PhonerLite] >>> variable_sip_via_host: [192.168.1.104] >>> variable_sip_via_port: [5060] >>> variable_bypass_media: [true] >>> variable_proxy_media: [true] >>> variable_sip_via_rport: [5060] >>> variable_max_forwards: [70] >>> variable_switch_r_sdp: [v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> ] >>> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >>> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >>> variable_effective_caller_id_number: [PhonerLite] >>> variable_effective_caller_id_name: [PhonerLite] >>> variable_>> variable_routing_digit: [90964111] >>> variable_continue_on_fail: [true] >>> variable_hangup_after_bridge: [true] >>> variable_sip_contact_user: [PhonerLite] >>> >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 Patched SDP >>> --- >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> +++ >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >>> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >>> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send signal >>> sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to >>> sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going >>> to sleep >>> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >>> Max-Forwards: 69 >>> From: "PhonerLite" >>> >;tag=8tH6Xjt2XaU9F >>> To: >>> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >>> CSeq: 123735760 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 404 >>> Remote-Party-ID: "PhonerLite" >>> >;party=calling;screen=yes;privacy=off >>> >>> v=0 >>> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/db91a3af/attachment-0002.html -------------- next part -------------- freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> recv 951 bytes from udp/[192.168.1.104]:5060 at 11:49:56.930364: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport From: ;tag=2563216860 To: Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces User-Agent: SIPPER for PhonerLite Content-Length: 396 v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv ------------------------------------------------------------------------ send 351 bytes to udp/[192.168.1.104]:5060 at 11:49:56.930781: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.930444 [NOTICE] switch_channel.c:613 New Channel sofia/internal/PhonerLite at 192.168.1.102 [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [received][100] 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2009-12-02 19:49:56.930444 [DEBUG] sofia.c:3490 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_NEW -> CS_INIT 2009-12-02 19:49:56.930444 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_INIT 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/PhonerLite at 192.168.1.102) State INIT 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:83 sofia/internal/PhonerLite at 192.168.1.102 SOFIA INIT 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:111 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/PhonerLite at 192.168.1.102) State INIT going to sleep 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/PhonerLite at 192.168.1.102) State ROUTING 2009-12-02 19:49:56.931815 [DEBUG] mod_sofia.c:130 sofia/internal/PhonerLite at 192.168.1.102 SOFIA ROUTING 2009-12-02 19:49:56.931815 [DEBUG] switch_core_state_machine.c:78 sofia/internal/PhonerLite at 192.168.1.102 Standard ROUTING 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_ROUTING -> CS_EXECUTE 2009-12-02 19:49:56.946388 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/PhonerLite at 192.168.1.102) State ROUTING going to sleep 2009-12-02 19:49:56.946388 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/PhonerLite at 192.168.1.102) State EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] mod_sofia.c:173 sofia/internal/PhonerLite at 192.168.1.102 SOFIA EXECUTE 2009-12-02 19:49:56.947392 [DEBUG] switch_core_state_machine.c:151 sofia/internal/PhonerLite at 192.168.1.102 Standard EXECUTE EXECUTE sofia/internal/PhonerLite at 192.168.1.102 set(sip_contact_user=PhonerLite) EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 19:49:56.957384 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_INIT 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA INIT 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to sleep 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING 2009-12-02 19:49:56.959386 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA ROUTING 2009-12-02 19:49:56.959386 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going to sleep 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA 2009-12-02 19:49:56.959386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA going to sleep send 1156 bytes to udp/[192.168.1.116]:9390 at 11:49:56.960484: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m Max-Forwards: 69 From: "PhonerLite" ;tag=r0pv05c0848ae To: Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 401 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 523740055483911509 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:56.960392 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9390 entering state [calling][0] recv 342 bytes from udp/[192.168.1.116]:9390 at 11:49:56.962288: ------------------------------------------------------------------------ SIP/2.0 403 Because Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m From: "PhonerLite" ;tag=r0pv05c0848ae To: ;tag=6 Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 332 bytes to udp/[192.168.1.116]:9390 at 11:49:56.962463: ------------------------------------------------------------------------ ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m Max-Forwards: 69 From: "PhonerLite" ;tag=r0pv05c0848ae To: ;tag=6 Call-ID: a76ed230-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.962396 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9390 entering state [terminated][403] 2009-12-02 19:49:56.962396 [NOTICE] sofia.c:3925 Hangup sofia/internal/90964111 at 192.168.1.116:9390 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2009-12-02 19:49:56.962396 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [KILL] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_HANGUP 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9390) State HANGUP 2009-12-02 19:49:56.962396 [DEBUG] mod_sofia.c:306 sofia/internal/90964111 at 192.168.1.116:9390 Overriding SIP cause 603 with 403 from the other leg 2009-12-02 19:49:56.962396 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.116:9390 hanging up, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.116:9390 Standard HANGUP, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9390) State HANGUP going to sleep 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9390) State REPORTING 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.116:9390 Standard REPORTING, cause: CALL_REJECTED 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9390) State REPORTING going to sleep 2009-12-02 19:49:56.962396 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.116:9390) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] 2009-12-02 19:49:56.962396 [DEBUG] switch_core_session.c:1069 Session 11 (sofia/internal/90964111 at 192.168.1.116:9390) Locked, Waiting on external entities 2009-12-02 19:49:56.963391 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2009-12-02 19:49:56.963391 [INFO] mod_dptools.c:2106 Originate Failed. Cause: CALL_REJECTED 2009-12-02 19:49:56.963391 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: CALL_REJECTED 2009-12-02 19:49:56.963391 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.116:9390] Other-Leg-Unique-ID: [aca42f97-803b-40b4-93b5-79531cdd47e7] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.116:9390] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:56.973380 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [NOTICE] switch_core_session.c:1087 Session 11 (sofia/internal/90964111 at 192.168.1.116:9390) Ended 2009-12-02 19:49:56.974397 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.116:9390 [CS_DESTROY] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change CS_DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9390) State DESTROY 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.116:9390 SOFIA DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.116:9390 Standard DESTROY 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9390) State DESTROY going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_INIT 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9391) State INIT 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA INIT 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.116:9391) State INIT going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9391) State ROUTING 2009-12-02 19:49:56.974397 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA ROUTING 2009-12-02 19:49:56.974397 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.116:9391) State ROUTING going to sleep 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9391) State CONSUME_MEDIA 2009-12-02 19:49:56.974397 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.116:9391) State CONSUME_MEDIA going to sleep send 1157 bytes to udp/[192.168.1.116]:9391 at 11:49:56.980816: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg Max-Forwards: 69 From: "PhonerLite" ;tag=S9FN20X35DZXS To: Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 5392773558290384508 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:56.980394 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9391 entering state [calling][0] recv 342 bytes from udp/[192.168.1.116]:9391 at 11:49:56.990534: ------------------------------------------------------------------------ SIP/2.0 403 Because Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg From: "PhonerLite" ;tag=S9FN20X35DZXS To: ;tag=6 Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 332 bytes to udp/[192.168.1.116]:9391 at 11:49:56.990729: ------------------------------------------------------------------------ ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg Max-Forwards: 69 From: "PhonerLite" ;tag=S9FN20X35DZXS To: ;tag=6 Call-ID: a771eccc-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:56.990410 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.116:9391 entering state [terminated][403] 2009-12-02 19:49:56.990410 [NOTICE] sofia.c:3925 Hangup sofia/internal/90964111 at 192.168.1.116:9391 [CS_CONSUME_MEDIA] [CALL_REJECTED] 2009-12-02 19:49:56.990410 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [KILL] 2009-12-02 19:49:56.990410 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.990410 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:49:56.990410 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_HANGUP 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9391) State HANGUP 2009-12-02 19:49:56.991392 [DEBUG] mod_sofia.c:306 sofia/internal/90964111 at 192.168.1.116:9391 Overriding SIP cause 603 with 403 from the other leg 2009-12-02 19:49:56.991392 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.116:9391 hanging up, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.116:9391 Standard HANGUP, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.116:9391) State HANGUP going to sleep 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9391) State REPORTING 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.116:9391 Standard REPORTING, cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.116:9391) State REPORTING going to sleep 2009-12-02 19:49:56.991392 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.116:9391) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.116:9391 [BREAK] 2009-12-02 19:49:56.991392 [DEBUG] switch_core_session.c:1069 Session 12 (sofia/internal/90964111 at 192.168.1.116:9391) Locked, Waiting on external entities 2009-12-02 19:49:56.991392 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2009-12-02 19:49:56.991392 [INFO] mod_dptools.c:2106 Originate Failed. Cause: CALL_REJECTED 2009-12-02 19:49:56.991392 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: CALL_REJECTED 2009-12-02 19:49:56.992388 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [ringing] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.116:9391] Other-Leg-Unique-ID: [990cc7dc-e7ea-4b82-850c-9c8c254136f9] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.116:9391] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_endpoint_disposition: [RECEIVED_NOMEDIA] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_ CS_INIT 2009-12-02 19:49:57.003383 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [NOTICE] switch_core_session.c:1087 Session 12 (sofia/internal/90964111 at 192.168.1.116:9391) Ended 2009-12-02 19:49:57.004386 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.116:9391 [CS_DESTROY] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.116:9391) Running State Change CS_DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9391) State DESTROY 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.116:9391 SOFIA DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.116:9391 Standard DESTROY 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.116:9391) State DESTROY going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_INIT 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.13:5060) State INIT 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:83 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA INIT 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:111 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_INIT -> CS_ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/90964111 at 192.168.1.13:5060) State INIT going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.13:5060) State ROUTING 2009-12-02 19:49:57.004386 [DEBUG] mod_sofia.c:130 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA ROUTING 2009-12-02 19:49:57.004386 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/90964111 at 192.168.1.13:5060) State ROUTING going to sleep 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA 2009-12-02 19:49:57.004386 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA going to sleep send 1155 bytes to udp/[192.168.1.13]:5060 at 11:49:57.012792: ------------------------------------------------------------------------ INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 402 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522444 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ 2009-12-02 19:49:57.012392 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [calling][0] recv 283 bytes from udp/[192.168.1.13]:5060 at 11:49:57.219574: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Content-Length: 0 ------------------------------------------------------------------------ recv 333 bytes from udp/[192.168.1.13]:5060 at 11:49:57.234550: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [proceeding][180] 2009-12-02 19:49:57.233383 [NOTICE] sofia.c:3423 Ring-Ready sofia/internal/90964111 at 192.168.1.13:5060! 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3432 sofia/internal/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 19:49:57.233383 [NOTICE] mod_sofia.c:1461 Ring-Ready sofia/internal/PhonerLite at 192.168.1.102! 2009-12-02 19:49:57.233383 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] send 618 bytes to udp/[192.168.1.104]:5060 at 11:49:57.235092: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:57.233383 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [early][180] 2009-12-02 19:49:57.233383 [DEBUG] switch_ivr_originate.c:1931 sofia/internal/PhonerLite at 192.168.1.102 receive message [RINGING] 2009-12-02 19:49:57.233383 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:57.233383 [NOTICE] switch_ivr_originate.c:1931 Ring Ready sofia/internal/PhonerLite at 192.168.1.102! recv 725 bytes from udp/[192.168.1.13]:5060 at 11:49:59.906802: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 INVITE Contact: Supported: replaces Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 256 v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 376 bytes to udp/[192.168.1.13]:5060 at 11:49:59.907203: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ Max-Forwards: 70 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758962 ACK Contact: Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [ready][200] 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-02 19:49:59.907094 [DEBUG] switch_channel.c:1935 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907094 [NOTICE] sofia.c:3834 Channel [sofia/internal/90964111 at 192.168.1.13:5060] has been answered 2009-12-02 19:49:59.907094 [DEBUG] switch_channel.c:1971 sofia/internal/90964111 at 192.168.1.13:5060 execute on answer: incre_call_stat(203 621 201 256 25 2591585 1) EXECUTE sofia/internal/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1) [1259754599] 2009-12-02 19:49:59.907094 [DEBUG] sofia.c:3847 sofia/internal/PhonerLite at 192.168.1.102 receive message [ANSWER] 2009-12-02 19:49:59.907094 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907911 [NOTICE] sofia.c:3847 Channel [sofia/internal/PhonerLite at 192.168.1.102] has been answered 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/internal/90964111 at 192.168.1.13:5060] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_bridge.c:806 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr_bridge.c:807 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] send 858 bytes to udp/[192.168.1.104]:5060 at 11:49:59.908359: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060 From: ;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 v=0 o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13 s=A conversation c=IN IP4 192.168.1.13 t=0 0 m=audio 10098 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ EXECUTE sofia/internal/PhonerLite at 192.168.1.102 get_next_route() 2009-12-02 19:49:59.907911 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [completed][200] EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() 2009-12-02 19:49:59.907911 [INFO] mod_dptools.c:955 CHANNEL_DATA: Channel-State: [CS_HIBERNATE] Channel-State-Number: [8] Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [answered] Caller-Username: [PhonerLite] Caller-Dialplan: [class4] Caller-Caller-ID-Name: [PhonerLite] Caller-Caller-ID-Number: [PhonerLite] Caller-Network-Addr: [192.168.1.104] Caller-Destination-Number: [90964111] Caller-Unique-ID: [85042f75-de46-4fe5-b8d8-53e4ca0d26e1] Caller-Source: [mod_sofia] Caller-Context: [default] Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] Caller-Profile-Index: [1] Caller-Profile-Created-Time: [1259754596930444] Caller-Channel-Created-Time: [1259754596930444] Caller-Channel-Answered-Time: [1259754599907094] Caller-Channel-Progress-Time: [1259754597233383] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] Other-Leg-Username: [PhonerLite] Other-Leg-Caller-ID-Name: [PhonerLite] Other-Leg-Caller-ID-Number: [PhonerLite] Other-Leg-Network-Addr: [192.168.1.104] Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] Other-Leg-Unique-ID: [e73fbd5b-a2ca-4685-bc6e-e1f848e8c7e5] Other-Leg-Source: [mod_sofia] Other-Leg-Channel-Name: [sofia/internal/90964111 at 192.168.1.13:5060] Other-Leg-Screen-Bit: [true] Other-Leg-Privacy-Hide-Name: [false] Other-Leg-Privacy-Hide-Number: [false] variable_sip_received_ip: [192.168.1.104] variable_sip_received_port: [5060] variable_sip_via_protocol: [udp] variable_sip_from_user: [PhonerLite] variable_sip_from_uri: [PhonerLite at 192.168.1.102] variable_sip_from_host: [192.168.1.102] variable_sip_from_user_stripped: [PhonerLite] variable_sip_from_tag: [2563216860] variable_sofia_profile_name: [internal] variable_sip_req_user: [90964111] variable_sip_req_uri: [90964111 at 192.168.1.102] variable_sip_req_host: [192.168.1.102] variable_sip_to_user: [90964111] variable_sip_to_uri: [90964111 at 192.168.1.102] variable_sip_to_host: [192.168.1.102] variable_sip_contact_port: [5060] variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] variable_sip_contact_host: [192.168.1.104] variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] variable_sip_call_id: [80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104] variable_sip_user_agent: [SIPPER for PhonerLite] variable_sip_via_host: [192.168.1.104] variable_sip_via_port: [5060] variable_sip_via_rport: [5060] variable_max_forwards: [70] variable_switch_r_sdp: [v=0 o=- 478760567 0 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.104 t=0 0 m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ] variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] variable_effective_caller_id_number: [PhonerLite] variable_effective_caller_id_name: [PhonerLite] variable_;tag=2563216860 To: ;tag=QQX3yavvBvjrj Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 17 ACK Contact: Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:49:59.907911 [DEBUG] mod_dptools.c:752 sofia/internal/PhonerLite at 192.168.1.102 SET [egress_alias]=[9392] 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:355 (sofia/internal/90964111 at 192.168.1.13:5060) State HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] mod_sofia.c:160 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:212 sofia/internal/90964111 at 192.168.1.13:5060 Standard HIBERNATE 2009-12-02 19:49:59.907911 [DEBUG] switch_core_state_machine.c:355 (sofia/internal/90964111 at 192.168.1.13:5060) State HIBERNATE going to sleep 2009-12-02 19:49:59.907911 [DEBUG] sofia.c:3359 Channel sofia/internal/PhonerLite at 192.168.1.102 entering state [ready][200] 2009-12-02 19:49:59.907911 [DEBUG] switch_ivr.c:1159 sofia/internal/PhonerLite at 192.168.1.102 receive message [MEDIA] 2009-12-02 19:49:59.917033 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:49:59.917033 [CRIT] switch_core_io.c:115 sofia/internal/PhonerLite at 192.168.1.102 reading on a session with no media! 2009-12-02 19:49:59.918616 [DEBUG] switch_ivr.c:1174 sofia/internal/90964111 at 192.168.1.13:5060 receive message [MEDIA] 2009-12-02 19:49:59.918616 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] send 955 bytes to udp/[192.168.1.13]:5060 at 11:49:59.922755: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522445 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 33352 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ 2009-12-02 19:49:59.922846 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [calling][0] send 955 bytes to udp/[192.168.1.13]:5060 at 11:50:00.423821: ------------------------------------------------------------------------ INVITE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 223 Remote-Party-ID: "PhonerLite" ;party=calling;screen=yes;privacy=off v=0 o=- 478760567 3248233194293522445 IN IP4 192.168.1.104 s=SIPPER for PhonerLite c=IN IP4 192.168.1.102 t=0 0 m=audio 33352 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ recv 337 bytes from udp/[192.168.1.13]:5060 at 11:50:00.444374: ------------------------------------------------------------------------ SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 330 bytes to udp/[192.168.1.13]:5060 at 11:50:00.444561: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-12-02 19:50:00.445918 [DEBUG] sofia.c:3359 Channel sofia/internal/90964111 at 192.168.1.13:5060 entering state [ready][400] 2009-12-02 19:50:00.445918 [NOTICE] sofia.c:3891 Hangup sofia/internal/90964111 at 192.168.1.13:5060 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.445918 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [KILL] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_HANGUP 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.13:5060) State HANGUP 2009-12-02 19:50:00.445918 [DEBUG] mod_sofia.c:338 Channel sofia/internal/90964111 at 192.168.1.13:5060 hanging up, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/internal/90964111 at 192.168.1.13:5060 2009-12-02 19:50:00.445918 [NOTICE] switch_ivr_bridge.c:727 Hangup sofia/internal/PhonerLite at 192.168.1.102 [CS_HIBERNATE] [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.445918 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/PhonerLite at 192.168.1.102 [KILL] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:46 sofia/internal/90964111 at 192.168.1.13:5060 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/90964111 at 192.168.1.13:5060) State HANGUP going to sleep 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:494 Hangup Command decre_call_stat(203 621 201 256 25 2591585 1): 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.13:5060) State REPORTING 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:53 sofia/internal/90964111 at 192.168.1.13:5060 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/90964111 at 192.168.1.13:5060) State REPORTING going to sleep 2009-12-02 19:50:00.445918 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/90964111 at 192.168.1.13:5060) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/90964111 at 192.168.1.13:5060 [BREAK] 2009-12-02 19:50:00.445918 [DEBUG] switch_core_session.c:1069 Session 13 (sofia/internal/90964111 at 192.168.1.13:5060) Locked, Waiting on external entities 2009-12-02 19:50:00.447416 [ERR] switch_core_io.c:120 sofia/internal/90964111 at 192.168.1.13:5060 has no read codec. 2009-12-02 19:50:00.447416 [DEBUG] switch_ivr_bridge.c:1210 originator uuid 85042f75-de46-4fe5-b8d8-53e4ca0d26e1 is not present 2009-12-02 19:50:00.447416 [NOTICE] switch_core_session.c:1087 Session 13 (sofia/internal/90964111 at 192.168.1.13:5060) Ended 2009-12-02 19:50:00.447416 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/90964111 at 192.168.1.13:5060 [CS_DESTROY] 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/90964111 at 192.168.1.13:5060) Running State Change CS_DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.13:5060) State DESTROY 2009-12-02 19:50:00.447416 [DEBUG] mod_sofia.c:255 sofia/internal/90964111 at 192.168.1.13:5060 SOFIA DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:60 sofia/internal/90964111 at 192.168.1.13:5060 Standard DESTROY 2009-12-02 19:50:00.447416 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/90964111 at 192.168.1.13:5060) State DESTROY going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_ivr_originate.c:2273 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2009-12-02 19:50:00.449755 [INFO] mod_dptools.c:2106 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] mod_dptools.c:2128 Continue on fail [true]: Cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:1371 Channel is hungup, aborting execution of application: get_next_route 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/PhonerLite at 192.168.1.102) State EXECUTE going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_HANGUP 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/PhonerLite at 192.168.1.102) State HANGUP 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:306 sofia/internal/PhonerLite at 192.168.1.102 Overriding SIP cause 488 with 403 from the other leg 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:338 Channel sofia/internal/PhonerLite at 192.168.1.102 hanging up, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/internal/PhonerLite at 192.168.1.102 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:46 sofia/internal/PhonerLite at 192.168.1.102 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/PhonerLite at 192.168.1.102) State HANGUP going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_HANGUP -> CS_REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/PhonerLite at 192.168.1.102) State REPORTING 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:53 sofia/internal/PhonerLite at 192.168.1.102 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:555 (sofia/internal/PhonerLite at 192.168.1.102) State REPORTING going to sleep 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/PhonerLite at 192.168.1.102) State Change CS_REPORTING -> CS_DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/PhonerLite at 192.168.1.102 [BREAK] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_session.c:1069 Session 10 (sofia/internal/PhonerLite at 192.168.1.102) Locked, Waiting on external entities 2009-12-02 19:50:00.449755 [NOTICE] switch_core_session.c:1087 Session 10 (sofia/internal/PhonerLite at 192.168.1.102) Ended 2009-12-02 19:50:00.449755 [NOTICE] switch_core_session.c:1089 Close Channel sofia/internal/PhonerLite at 192.168.1.102 [CS_DESTROY] 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/PhonerLite at 192.168.1.102) Running State Change CS_DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/PhonerLite at 192.168.1.102) State DESTROY 2009-12-02 19:50:00.449755 [DEBUG] mod_sofia.c:255 sofia/internal/PhonerLite at 192.168.1.102 SOFIA DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:60 sofia/internal/PhonerLite at 192.168.1.102 Standard DESTROY 2009-12-02 19:50:00.449755 [DEBUG] switch_core_state_machine.c:412 (sofia/internal/PhonerLite at 192.168.1.102) State DESTROY going to sleep send 620 bytes to udp/[192.168.1.13]:5060 at 11:50:00.452239: ------------------------------------------------------------------------ BYE sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e Max-Forwards: 70 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758964 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ send 641 bytes to udp/[192.168.1.104]:5060 at 11:50:00.452431: ------------------------------------------------------------------------ BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma Max-Forwards: 70 From: ;tag=QQX3yavvBvjrj To: ;tag=2563216860 Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 123758964 BYE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ recv 376 bytes from udp/[192.168.1.104]:5060 at 11:50:00.453188: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma From: ;tag=QQX3yavvBvjrj To: ;tag=2563216860 Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104 CSeq: 123758964 BYE Contact: User-Agent: SIPPER for PhonerLite Content-Length: 0 ------------------------------------------------------------------------ recv 337 bytes from udp/[192.168.1.13]:5060 at 11:50:00.530378: ------------------------------------------------------------------------ SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 INVITE Contact: Content-Length: 0 ------------------------------------------------------------------------ send 330 bytes to udp/[192.168.1.13]:5060 at 11:50:00.530547: ------------------------------------------------------------------------ ACK sip:192.168.1.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK Max-Forwards: 69 From: "PhonerLite" ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758963 ACK Content-Length: 0 ------------------------------------------------------------------------ recv 325 bytes from udp/[192.168.1.13]:5060 at 11:50:00.821952: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e From: PhonerLite ;tag=tj9D4Ue72pNgN To: ;tag=2900712955 Call-ID: a776ce22-59db-122d-ff84-0022190e9476 CSeq: 123758964 BYE Contact: Content-Length: 0 ------------------------------------------------------------------------ From mike at jerris.com Tue Dec 1 20:26:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 1 Dec 2009 23:26:22 -0500 Subject: [Freeswitch-users] freeswitch changes c= to FS's IP with bypass_media=true In-Reply-To: <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> References: <27c25bc40912010559o439ed435s133e5348f627131f@mail.gmail.com> <27c25bc40912010708q56114dddvcee3f75a9813f77e@mail.gmail.com> <191c3a030912011159i65c41fb2gf2c8758b221fcf41@mail.gmail.com> <27c25bc40912011911w38387563h4f469ed4304ae3f8@mail.gmail.com> <27c25bc40912012007g6d860261v9a4ae50a77db47eb@mail.gmail.com> Message-ID: <944399F2-5F9F-437D-B57B-5DA4FC472B76@jerris.com> That is not a freeswitch log and the call path you describe is not what was in the previous freeswitch log you posted. Please post the complete freeswitch debug log with siptrace enabled for that callflow you described. On Dec 1, 2009, at 11:07 PM, Juan Backson wrote: > Hi Mike, > > > Here is a very strange SIP outgoing INVITES from freeswitch: > > The call path is this: > > 192.168.1.104 (phonerlite) -> 192.168.1.102 ( freeswitch ) > 192.168.1.102 -> 192.168.1.116 (sipp gives back 403) > 192.168.1.102 -> 192.168.1.13 ( phone ) > > The first INVITE to 192.168.1.13 has the right c= and o= ( both is > pointing to 192.168.1.104). But the for some unknown reason, > Freeswitch sends INVITES again. But in the INVITE resend, the o = > becomes fs's ip. > > I have no idea. This is only bypass_meida. > > Attached is the fs log. > > > ngrep -q -p -W byline port 5060 > interface: eth0 (192.168.1.0/255.255.255.0) > filter: (ip) and ( port 5060 ) > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > INVITE sip:90964111 at 192.168.1.102 SIP/2.0. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport. > From: ;tag=2563216860. > To: . > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > Content-Type: application/sdp. > Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, > UPDATE. > Max-Forwards: 70. > Supported: 100rel, replaces. > User-Agent: SIPPER for PhonerLite. > Content-Length: 396. > . > v=0. > o=- 478760567 0 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=sendrecv. > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: . > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9390 > INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > Max-Forwards: 69. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: . > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 401. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 523740055483911509 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.116:9390 -> 192.168.1.102:5060 > SIP/2.0 403 Because. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: ;tag=6. > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9390 > ACK sip:90964111 at 192.168.1.116:9390 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKS9Qm26ZS4U93m. > Max-Forwards: 69. > From: "PhonerLite" ;tag=r0pv05c0848ae. > To: ;tag=6. > Call-ID: a76ed230-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9391 > INVITE sip:90964111 at 192.168.1.116:9391 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > Max-Forwards: 69. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: . > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 402. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 5392773558290384508 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.116:9391 -> 192.168.1.102:5060 > SIP/2.0 403 Because. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: ;tag=6. > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.116:9391 > ACK sip:90964111 at 192.168.1.116:9391 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKtjHD41gX14Zpg. > Max-Forwards: 69. > From: "PhonerLite" ;tag=S9FN20X35DZXS. > To: ;tag=6. > Call-ID: a771eccc-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: . > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 402. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522444 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.104. > t=0 0. > m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:2 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:97 iLBC/8000. > a=rtpmap:110 speex/8000. > a=rtpmap:111 speex/16000. > a=rtpmap:9 G722/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: . > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKUUa65v10yDp9B. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 INVITE. > Contact: . > Supported: replaces. > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, > PRACK, UPDATE, MESSAGE. > Content-Type: application/sdp. > Content-Length: 256. > . > v=0. > o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. > s=A conversation. > c=IN IP4 192.168.1.13. > t=0 0. > m=audio 10098 RTP/AVP 8 0 9 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:9 G722/16000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > ACK sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKv43y7Qj4UpcvQ. > Max-Forwards: 70. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758962 ACK. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK80f2b42464ddde1180d3001a805656a5;rport=5060. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 207. > . > v=0. > o=sdp_admin 59711527 21326134 IN IP4 192.168.1.13. > s=A conversation. > c=IN IP4 192.168.1.13. > t=0 0. > m=audio 10098 RTP/AVP 8 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0. > Via: SIP/2.0/UDP > 192.168.1.104: > 5060;branch=z9hG4bK00b67e2664ddde1180d3001a805656a5;rport. > From: ;tag=2563216860. > To: ;tag=QQX3yavvBvjrj. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 17 ACK. > Contact: . > Max-Forwards: 70. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 223. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.102. > t=0 0. > m=audio 33352 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > > > U 192.168.1.104:43488 -> 192.168.1.102:5060 > . > . > .............. > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > INVITE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 223. > Remote-Party-ID: "PhonerLite" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=- 478760567 3248233194293522445 IN IP4 192.168.1.104. > s=SIPPER for PhonerLite. > c=IN IP4 192.168.1.102. > t=0 0. > m=audio 33352 RTP/AVP 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 400 Bad Request. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > ACK sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > Max-Forwards: 69. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 ACK. > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.13:5060 > BYE sip:192.168.1.13:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKyppgBemBp8r1e. > Max-Forwards: 70. > From: "PhonerLite" ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758964 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". > Content-Length: 0. > . > > > U 192.168.1.102:5060 -> 192.168.1.104:5060 > BYE sip:PhonerLite at 192.168.1.104:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKZZF9c94eKHFma. > Max-Forwards: 70. > From: ;tag=QQX3yavvBvjrj. > To: ;tag=2563216860. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 123758964 BYE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION". > Content-Length: 0. > . > > > U 192.168.1.104:5060 -> 192.168.1.102:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.102;rport=5060;branch=z9hG4bKZZF9c94eKHFma. > From: ;tag=QQX3yavvBvjrj. > To: ;tag=2563216860. > Call-ID: 80F2B424-64DD-DE11-80D2-001A805656A5 at 192.168.1.104. > CSeq: 123758964 BYE. > Contact: . > User-Agent: SIPPER for PhonerLite. > Content-Length: 0. > . > > > U 192.168.1.13:5060 -> 192.168.1.102:5060 > SIP/2.0 400 Bad Request. > Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKXDXQ9j37rZ2eK. > From: PhonerLite ;tag=tj9D4Ue72pNgN. > To: ;tag=2900712955. > Call-ID: a776ce22-59db-122d-ff84-0022190e9476. > CSeq: 123758963 INVITE. > Contact: . > Content-Length: 0. > . > > > > On Wed, Dec 2, 2009 at 11:26 AM, Michael Jerris > wrote: > how are you sending both invites here? can you explain the full > call path and how you are originating these calls? > > > On Dec 1, 2009, at 10:11 PM, Juan Backson wrote: > >> Hi, >> >> I also did try to set?only bypass_media, but it still does not wor >> k??freeswitch still modifies the c= line, causing the call to fa >> il. >> >> Could someone please help? >> >> >> send 1155 bytes to udp/[192.168.1.13]:5060 at 10:56:57.516650: >> >> --- >> --------------------------------------------------------------------- >> INVITE sip:90964111 at 192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> Max-Forwards: 69 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 402 >> Remote-Party-ID: "PhonerLite" >> ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 794697697 5289748556544955553 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:111 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change CS_INIT -> >> CS_ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> 2009-12-02 18:56:57.516831 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:330 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State INIT going to >> sleep >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] mod_sofia.c:130 sofia/ >> external_1/90964111 at 192.168.1.13:5060 SOFIA ROUTING >> 2009-12-02 18:56:57.516831 [DEBUG] switch_ivr_originate.c:66 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change CS_ROUTING -> >> CS_CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:333 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State ROUTING going >> to sleep >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA >> 2009-12-02 18:56:57.516831 [DEBUG] switch_core_state_machine.c:352 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State CONSUME_MEDIA >> going to sleep >> recv 283 bytes from udp/[192.168.1.13]:5060 at 10:56:57.721536: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> recv 330 bytes from udp/[192.168.1.13]:5060 at 10:56:57.736450: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [proceeding] >> [180] >> 2009-12-02 18:56:57.736234 [NOTICE] sofia.c:3423 Ring-Ready sofia/ >> external_1/90964111 at 192.168.1.13:5060! >> 2009-12-02 18:56:57.736234 [DEBUG] sofia.c:3432 sofia/external_1/ >> PhonerLite at 192.168.1.102 receive message [RINGING] >> 2009-12-02 18:56:57.736234 [NOTICE] mod_sofia.c:1461 Ring-Ready >> sofia/external_1/PhonerLite at 192.168.1.102! >> 2009-12-02 18:56:57.736234 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> send 618 bytes to udp/[192.168.1.104]:5060 at 10:56:57.737121: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:57.737317 [DEBUG] switch_ivr_originate.c:1931 >> sofia/external_1/PhonerLite at 192.168.1.102 receive message [RINGING] >> 2009-12-02 18:56:57.737317 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [early][180] >> 2009-12-02 18:56:57.737317 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:57.737317 [NOTICE] switch_ivr_originate.c:1931 >> Ring Ready sofia/external_1/PhonerLite at 192.168.1.102! >> recv 722 bytes from udp/[192.168.1.13]:5060 at 10:56:59.381338: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKmvSrSZB8jFt6a >> From: PhonerLite ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 INVITE >> Contact: >> Supported: replaces >> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, >> PRACK, UPDATE, MESSAGE >> Content-Type: application/sdp >> Content-Length: 256 >> >> v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> --- >> --------------------------------------------------------------------- >> send 373 bytes to udp/[192.168.1.13]:5060 at 10:56:59.381739: >> >> --- >> --------------------------------------------------------------------- >> ACK sip:192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKN5jHUtvBgrgSp >> Max-Forwards: 70 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757372 ACK >> Contact: >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [ready][200] >> 2009-12-02 18:56:59.381828 [DEBUG] sofia.c:3366 Remote SDP: >> v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1935 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:59.381828 [NOTICE] sofia.c:3834 Channel [sofia/ >> external_1/90964111 at 192.168.1.13:5060] has been answered >> 2009-12-02 18:56:59.381828 [DEBUG] switch_channel.c:1971 sofia/ >> external_1/90964111 at 192.168.1.13:5060 execute on answer: >> incre_call_stat(203 621 201 256 25 2591585 1) >> EXECUTE sofia/external_1/90964111 at 192.168.1.13:5060 incre_call_stat(203 621 201 256 25 2591585 1 >> ) >> >> 2009-12-02 18:56:59.382721 [NOTICE] switch_ivr_originate.c:2152 >> Channel [sofia/external_1/PhonerLite at 192.168.1.102] has been answered >> 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_originate.c:2196 >> Originate Resulted in Success: [sofia/ >> external_1/90964111 at 192.168.1.13:5060] >> send 858 bytes to udp/[192.168.1.104]:5060 at 10:56:59.382955: >> >> --- >> --------------------------------------------------------------------- >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK000be0bd5cddde1180d2001a805656a5;rport=5060 >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N2009-12-02 >> 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:806 (sofia/external_1/ >> PhonerLite at 192.168.1.102) State Change CS_EXECUTE -> CS_HIBERNATE >> >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:474 bypass_media= >> [true] >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> 2009-12-02 18:56:59.382721 [DEBUG] mod_limit.c:486 >> originate_disposition=[SUCCESS] >> Content-Type: application/sdp >> Content-Disposition: session >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> Content-Length: 207 >> >> v=0 >> 2009-12-02 18:56:59.382721 [DEBUG] switch_ivr_bridge.c:807 (sofia/ >> external_1/90964111 at 192.168.1.13:5060) State Change >> CS_CONSUME_MEDIA -> CS_HIBERNATE >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> m=audio 10096 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:306 >> (sofia/external_1/90964111 at 192.168.1.13:5060) Running State Change >> CS_HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] mod_sofia.c:160 sofia/ >> external_1/90964111 at 192.168.1.13:5060 SOFIA HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:212 >> sofia/external_1/90964111 at 192.168.1.13:5060 Standard HIBERNATE >> 2009-12-02 18:56:59.382721 [DEBUG] switch_core_state_machine.c:355 >> (sofia/external_1/90964111 at 192.168.1.13:5060) State HIBERNATE going >> to sleep >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 get_next_route() >> 2009-12-02 18:56:59.382721 [DEBUG] mod_class4.c:2458 Starting to >> get next route... >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 info() >> 2009-12-02 18:56:59.383439 [INFO] mod_dptools.c:955 CHANNEL_DATA: >> Channel-State: [CS_HIBERNATE] >> Channel-State-Number: [8] >> Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] >> Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] >> Call-Direction: [inbound] >> Presence-Call-Direction: [inbound] >> Answer-State: [answered] >> Caller-Username: [PhonerLite] >> Caller-Dialplan: [class4] >> Caller-Caller-ID-Name: [PhonerLite] >> Caller-Caller-ID-Number: [PhonerLite] >> Caller-Network-Addr: [192.168.1.104] >> Caller-Destination-Number: [90964111] >> Caller-Unique-ID: [01149fda-c473-4c17-859f-66c016b1e376] >> Caller-Source: [mod_sofia] >> Caller-Context: [default] >> Caller-Channel-Name: [sofia/external_1/PhonerLite at 192.168.1.102] >> Caller-Profile-Index: [1] >> Caller-Profile-Created-Time: [1259751417450336] >> Caller-Channel-Created-Time: [1259751417450336] >> Caller-Channel-Answered-Time: [1259751419381828] >> Caller-Channel-Progress-Time: [1259751417736234] >> Caller-Channel-Progress-Media-Time: [0] >> Caller-Channel-Hangup-Time: [0] >> Caller-Channel-Transfer-Time: [0] >> Caller-Screen-Bit: [true] >> Caller-Privacy-Hide-Name: [false] >> Caller-Privacy-Hide-Number: [false] >> Other-Leg-Username: [PhonerLite] >> Other-Leg-Caller-ID-Name: [PhonerLite] >> Other-Leg-Caller-ID-Number: [PhonerLite] >> Other-Leg-Network-Addr: [192.168.1.104] >> Other-Leg-Destination-Number: [90964111 at 192.168.1.13:5060] >> Other-Leg-Unique-ID: [4db93c42-909f-4299-96a6-416335744dbe] >> Other-Leg-Source: [mod_sofia] >> Other-Leg-Channel-Name: [sofia/external_1/90964111 at 192.168.1.13:5060] >> Other-Leg-Screen-Bit: [true] >> Other-Leg-Privacy-Hide-Name: [false] >> Other-Leg-Privacy-Hide-Number: [false] >> variable_sip_received_ip: [192.168.1.104] >> variable_sip_received_port: [5060] >> variable_sip_via_protocol: [udp] >> variable_sip_from_user: [PhonerLite] >> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >> variable_sip_from_host: [192.168.1.102] >> variable_sip_from_user_stripped: [PhonerLite] >> variable_sip_from_tag: [2454193703] >> variable_sofia_profile_name: [external_1] >> variable_sip_req_user: [90964111] >> variable_sip_req_uri: [90964111 at 192.168.1.102] >> variable_sip_req_host: [192.168.1.102] >> variable_sip_to_user: [90964111] >> variable_sip_to_uri: [90964111 at 192.168.1.102] >> variable_sip_to_host: [192.168.1.102] >> variable_sip_contact_port: [5060] >> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >> variable_sip_contact_host: [192.168.1.104] >> variable_channel_name: [sofia/external_1/PhonerLite at 192.168.1.102] >> variable_sip_call_id: [000BE0BD-5CDD- >> DE11-80D1-001A805656A5 at 192.168.1.104] >> variable_sip_user_agent: [SIPPER for PhonerLite] >> variable_sip_via_host: [192.168.1.104] >> variable_sip_via_port: [5060] >> variable_sip_via_rport: [5060] >> variable_max_forwards: [70] >> variable_switch_r_sdp: [v=0 >> o=- 794697697 0 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.104 >> t=0 0 >> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:97 iLBC/8000 >> a=rtpmap:110 speex/8000 >> a=rtpmap:111 speex/16000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> ] >> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >> variable_effective_caller_id_number: [PhonerLite] >> variable_effective_caller_id_name: [PhonerLite] >> variable_> variable_routing_digit: [90964111] >> variable_continue_on_fail: [true] >> variable_hangup_after_bridge: [true] >> variable_sip_contact_user: [PhonerLite] >> variable_proto_specific_hangup_cause: [sip:403] >> variable_sip_hangup_phrase: [Because] >> variable_bypass_media: [true] >> variable_success_bridge: [true] >> variable_signal_bond: [4db93c42-909f-4299-96a6-416335744dbe] >> variable_switch_m_sdp: [v=0 >> o=sdp_admin 11293200 10985190 IN IP4 192.168.1.13 >> s=A conversation >> c=IN IP4 192.168.1.13 >> t=0 0 >> m=audio 10096 RTP/AVP 8 0 9 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:9 G722/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> ] >> variable_endpoint_disposition: [ANSWER] >> variable_originate_disposition: [SUCCESS] >> variable_signal_bridge_to: [4db93c42-909f-4299-96a6-416335744dbe] >> variable_current_application: [info] >> >> recv 414 bytes from udp/[192.168.1.104]:5060 at 10:56:59.384444: >> >> --- >> --------------------------------------------------------------------- >> ACK sip:90964111 at 192.168.1.102:5060;transport=udp SIP/2.0 >> Via: SIP/2.0/UDP >> 192.168.1.104: >> 5060;branch=z9hG4bK003811bf5cddde1180d2001a805656a5;rport >> From: ;tag=2454193703 >> To: ;tag=FFKXgjN02m02N >> Call-ID: 000BE0BD-5CDD-DE11-80D1-001A805656A5 at 192.168.1.104 >> CSeq: 15 ACK2009-12-02 18:56:59.383439 [DEBUG] mod_dptools.c:752 >> sofia/external_1/PhonerLite at 192.168.1.102 SET [final_digits]= >> [90964111] >> >> Contact: >> Max-Forwards: 70 >> Content-Length: 0 >> >> >> --- >> --------------------------------------------------------------------- >> >> EXECUTE sofia/external_1/PhonerLite at 192.168.1.102 bridge(sofia/ >> external_1/90964111 at 192.168.1.116:9392) >> 2009-12-02 18:56:59.390427 [DEBUG] switch_ivr.c:1159 sofia/ >> external_1/PhonerLite at 192.168.1.102 receive message [MEDIA] >> 2009-12-02 18:56:59.390427 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/PhonerLite at 192.168.1.102 [BREAK] >> 2009-12-02 18:56:59.390427 [CRIT] switch_core_io.c:115 sofia/ >> external_1/PhonerLite at 192.168.1.102 reading on a session with no >> media! >> 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [completed][200] >> 2009-12-02 18:56:59.390427 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/PhonerLite at 192.168.1.102 entering state [ready][200] >> 2009-12-02 18:56:59.393411 [DEBUG] switch_ivr.c:1174 sofia/ >> external_1/90964111 at 192.168.1.13:5060 receive message [MEDIA] >> 2009-12-02 18:56:59.393411 [DEBUG] switch_core_session.c:630 Send >> signal sofia/external_1/90964111 at 192.168.1.13:5060 [BREAK] >> send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.393770: >> >> --- >> --------------------------------------------------------------------- >> INVITE sip:192.168.1.13:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKpecaXNDFD16Bj >> Max-Forwards: 69 >> From: "PhonerLite" >> ;tag=jaZ7N37atF3tr >> To: ;tag=8849584 >> Call-ID: 40563247-59d4-122d-ff84-0022190e9476 >> CSeq: 123757373 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 223 >> Remote-Party-ID: "PhonerLite" >> ;party=calling;screen=yes;privacy=off >> >> v=0 >> o=- 794697697 5289748556544955554 IN IP4 192.168.1.104 >> s=SIPPER for PhonerLite >> c=IN IP4 192.168.1.102 >> t=0 0 >> m=audio 30632 RTP/AVP 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> >> --- >> --------------------------------------------------------------------- >> 2009-12-02 18:56:59.393411 [DEBUG] sofia.c:3359 Channel sofia/ >> external_1/90964111 at 192.168.1.13:5060 entering state [calling][0] >> send 952 bytes to udp/[192.168.1.13]:5060 at 10:56:59.894252: >> >> --- >> --------------------------------------------------------------------- >> >> On Wed, Dec 2, 2009 at 3:59 AM, Anthony Minessale > > wrote: >> yes he did you can see it in his trace. >> you can not use both of them together...... >> >> >> >> On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris >> wrote: >> The only way this would happen would be if this is set to proxy >> media not bypass. Are you setting both? >> >> Mike >> >> On Dec 1, 2009, at 10:08 AM, Juan Backson >> wrote: >> >>> In the following trace, 102 is FS IP, 104 is calling party and >>> 13 is called party. >>> >>> with bypass_media, FS still changes c=IN IP4 192.168.1.102 >>> >>> Any idea why? >>> >>> >>> freeswitch at localhost.localdomain> recv 951 bytes from udp/ >>> [192.168.1.104]:5060 at 22:56:33.782715: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.102 SIP/2.0 >>> Via: SIP/2.0/UDP >>> 192.168.1.104: >>> 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport >>> From: ;tag=786224322 >>> To: >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> Contact: >>> Content-Type: application/sdp >>> Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, >>> MESSAGE, UPDATE >>> Max-Forwards: 70 >>> Supported: 100rel, replaces >>> User-Agent: SIPPER for PhonerLite >>> Content-Length: 397 >>> >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=sendrecv >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> send 350 bytes to udp/[192.168.1.104]:5060 at 22:56:33.783145: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP >>> 192.168.1.104: >>> 5060;branch=z9hG4bK003c8e1bf8dcde11a854001a805656a5;rport=5060 >>> From: ;tag=786224322 >>> To: >>> Call-ID: 003C8E1B-F8DC-DE11-A853-001A805656A5 at 192.168.1.104 >>> CSeq: 37 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Content-Length: 0 >>> >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> 2009-12-02 06:56:33.783162 [NOTICE] switch_channel.c:613 New >>> Channel sofia/internal/PhonerLite at 192.168.1.102 [d4233c9a- >>> ee3b-40d4-910d-3b1579f9a273] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3359 Channel sofia/ >>> internal/PhonerLite at 192.168.1.102 entering state [received][100] >>> 2009-12-02 06:56:33.783162 [DEBUG] sofia.c:3366 Remote SDP: >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> >>> EXECUTE sofia/internal/PhonerLite at 192.168.1.102 info() >>> 2009-12-02 06:56:33.825198 [INFO] mod_dptools.c:955 CHANNEL_DATA: >>> Channel-State: [CS_EXECUTE] >>> Channel-State-Number: [4] >>> Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Call-Direction: [inbound] >>> Presence-Call-Direction: [inbound] >>> Answer-State: [ringing] >>> Caller-Username: [PhonerLite] >>> Caller-Dialplan: [class4] >>> Caller-Caller-ID-Name: [PhonerLite] >>> Caller-Caller-ID-Number: [PhonerLite] >>> Caller-Network-Addr: [192.168.1.104] >>> Caller-Destination-Number: [90964111] >>> Caller-Unique-ID: [d4233c9a-ee3b-40d4-910d-3b1579f9a273] >>> Caller-Source: [mod_sofia] >>> Caller-Context: [default] >>> Caller-Channel-Name: [sofia/internal/PhonerLite at 192.168.1.102] >>> Caller-Profile-Index: [1] >>> Caller-Profile-Created-Time: [1259708193783162] >>> Caller-Channel-Created-Time: [1259708193783162] >>> Caller-Channel-Answered-Time: [0] >>> Caller-Channel-Progress-Time: [0] >>> Caller-Channel-Progress-Media-Time: [0] >>> Caller-Channel-Hangup-Time: [0] >>> Caller-Channel-Transfer-Time: [0] >>> Caller-Screen-Bit: [true] >>> Caller-Privacy-Hide-Name: [false] >>> Caller-Privacy-Hide-Number: [false] >>> variable_sip_received_ip: [192.168.1.104] >>> variable_sip_received_port: [5060] >>> variable_sip_via_protocol: [udp] >>> variable_sip_from_user: [PhonerLite] >>> variable_sip_from_uri: [PhonerLite at 192.168.1.102] >>> variable_sip_from_host: [192.168.1.102] >>> variable_sip_from_user_stripped: [PhonerLite] >>> variable_sip_from_tag: [786224322] >>> variable_sofia_profile_name: [internal] >>> variable_sip_req_user: [90964111] >>> variable_sip_req_uri: [90964111 at 192.168.1.102] >>> variable_sip_req_host: [192.168.1.102] >>> variable_sip_to_user: [90964111] >>> variable_sip_to_uri: [90964111 at 192.168.1.102] >>> variable_sip_to_host: [192.168.1.102] >>> variable_sip_contact_port: [5060] >>> variable_sip_contact_uri: [PhonerLite at 192.168.1.104:5060] >>> variable_sip_contact_host: [192.168.1.104] >>> variable_channel_name: [sofia/internal/PhonerLite at 192.168.1.102] >>> variable_sip_call_id: [003C8E1B-F8DC-DE11- >>> A853-001A805656A5 at 192.168.1.104] >>> variable_sip_user_agent: [SIPPER for PhonerLite] >>> variable_sip_via_host: [192.168.1.104] >>> variable_sip_via_port: [5060] >>> variable_bypass_media: [true] >>> variable_proxy_media: [true] >>> variable_sip_via_rport: [5060] >>> variable_max_forwards: [70] >>> variable_switch_r_sdp: [v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> ] >>> variable_ep_codec_string: [PCMA at 8000h,PCMU at 8000h] >>> variable_endpoint_disposition: [RECEIVED_NOMEDIA] >>> variable_effective_caller_id_number: [PhonerLite] >>> variable_effective_caller_id_name: [PhonerLite] >>> variable_>> variable_routing_digit: [90964111] >>> variable_continue_on_fail: [true] >>> variable_hangup_after_bridge: [true] >>> variable_sip_contact_user: [PhonerLite] >>> >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] sofia_glue.c:1322 sofia/ >>> internal/90964111 at 192.168.1.116:9390 Patched SDP >>> --- >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.104 >>> t=0 0 >>> m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> +++ >>> v=0 >>> o=- 3393406017 0 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:111 (sofia/internal/ >>> 90964111 at 192.168.1.116:9390) State Change CS_INIT -> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:330 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State INIT going to >>> sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] mod_sofia.c:130 sofia/internal/ >>> 90964111 at 192.168.1.116:9390 SOFIA ROUTING >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_ivr_originate.c:66 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State Change >>> CS_ROUTING -> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_session.c:932 Send >>> signal sofia/internal/90964111 at 192.168.1.116:9390 [BREAK] >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State ROUTING going >>> to sleep >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:306 >>> (sofia/internal/90964111 at 192.168.1.116:9390) Running State Change >>> CS_CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> 2009-12-02 06:56:33.842639 [DEBUG] switch_core_state_machine.c:352 >>> (sofia/internal/90964111 at 192.168.1.116:9390) State CONSUME_MEDIA >>> going to sleep >>> send 1159 bytes to udp/[192.168.1.116]:9390 at 22:56:33.843976: >>> >>> --- >>> --- >>> ------------------------------------------------------------------ >>> INVITE sip:90964111 at 192.168.1.116:9390 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.102;rport;branch=z9hG4bKaNKUU3DjZjB1g >>> Max-Forwards: 69 >>> From: "PhonerLite" >>> ;tag=8tH6Xjt2XaU9F >>> To: >>> Call-ID: 9d052856-596f-122d-1b98-0022190e9476 >>> CSeq: 123735760 INVITE >>> Contact: >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15095 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, >>> INFO, REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 404 >>> Remote-Party-ID: "PhonerLite" >>> ;party=calling;screen=yes;privacy=off >>> >>> v=0 >>> o=- 3393406017 6776849011794265802 IN IP4 192.168.1.104 >>> s=SIPPER for PhonerLite >>> c=IN IP4 192.168.1.102 >>> t=0 0 >>> m=audio 34846 RTP/AVP 8 0 2 3 97 110 111 9 101 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:2 G726-32/8000 >>> a=rtpmap:3 GSM/8000 >>> a=rtpmap:97 iLBC/8000 >>> a=rtpmap:110 speex/8000 >>> a=rtpmap:111 speex/16000 >>> a=rtpmap:9 G722/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091201/2a6656ad/attachment-0002.html From max.clark at gmail.com Tue Dec 1 20:03:39 2009 From: max.clark at gmail.com (Max Clark) Date: Tue, 01 Dec 2009 20:03:39 -0800 Subject: [Freeswitch-users] OSP Interop w/ Trans Nexus In-Reply-To: <191c3a030810310758l4cf91940v39f30e611fd05e2c@mail.gmail.com> References: <200810311312.m9VDCoQM031972@omr12.networksolutionsemail.com> <191c3a030810310758l4cf91940v39f30e611fd05e2c@mail.gmail.com> Message-ID: Hi all, Did anything ever progress with this? Is there an option for OSP in FreeSWITCH? Thanks, Max On 10/31/08 7:58 AM, Anthony Minessale wrote: > We're here all the time if you want to collaborate on it. > We have 100+ users and developers in our irc channel and on this list so > it should not be an issue. > I'm sure we can find a few volunteers for testing. > > > On Fri, Oct 31, 2008 at 8:14 AM, Jim Dalton > > wrote: > > TransNexus would be glad to contribute the effort to add support for > the ETSI OSP protocol to Freeswitch if there is interest from the > community. Since we are not familiar with FreeSwitch, we will need > to collaborate with a FreeSwitch developer to understand how the OSP > client library (http://sourceforge.net/projects/osp-toolkit/) should > be integrated with Freeswitch. We will also need a user who can > perform beta testing on the Freeswitch OSP implementation. > Jim Dalton > VoIP Routing, Accounting, Security > 1.404.526.6053 > www.TransNexus.com From devel at thom.fr.eu.org Wed Dec 2 00:12:09 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 09:12:09 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> Message-ID: <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [2] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [4] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [5] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [6] http://www.freeswitch.org [7] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [8] ClueCon http://www.cluecon.com/ [9] Twitter: http://twitter.com/FreeSWITCH_wire [10] AIM: anthm MSN:anthony_minessale at hotmail.com [11] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [12] IRC: irc.freenode.net [13] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [14] iax:guest at conference.freeswitch.org/888 [15] googletalk:conf+888 at conference.freeswitch.org [16] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [17] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [18] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [19] http://www.freeswitch.org [20] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [21] ClueCon http://www.cluecon.com/ [22] Twitter: http://twitter.com/FreeSWITCH_wire [23] AIM: anthm MSN:anthony_minessale at hotmail.com [24] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [25] IRC: irc.freenode.net [26] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [27] iax:guest at conference.freeswitch.org/888 [28] googletalk:conf+888 at conference.freeswitch.org [29] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] http://files.freeswitch.org [3] mailto:devel at thom.fr.eu.org [4] mailto:FreeSWITCH-users at lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6] http://lists.freeswitch.org/mailman/options/freeswitch-users [7] http://www.freeswitch.org [8] http://www.freeswitch.org/ [9] http://www.cluecon.com/ [10] http://twitter.com/FreeSWITCH_wire [11] mailto:MSN%3Aanthony_minessale at hotmail.com [12] mailto:PAYPAL%3Aanthony.minessale at gmail.com [13] http://irc.freenode.net [14] mailto:sip%3A888 at conference.freeswitch.org [15] http://iax:guest at conference.freeswitch.org/888 [16] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [17] mailto:FreeSWITCH-users at lists.freeswitch.org [18] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19] http://lists.freeswitch.org/mailman/options/freeswitch-users [20] http://www.freeswitch.org [21] http://www.freeswitch.org/ [22] http://www.cluecon.com/ [23] http://twitter.com/FreeSWITCH_wire [24] mailto:MSN%3Aanthony_minessale at hotmail.com [25] mailto:PAYPAL%3Aanthony.minessale at gmail.com [26] http://irc.freenode.net [27] mailto:sip%3A888 at conference.freeswitch.org [28] http://iax:guest at conference.freeswitch.org/888 [29] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/77d63b4b/attachment-0002.html From abeka at greatiam.com Wed Dec 2 00:16:49 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 08:16:49 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers Message-ID: <4B162271.1010306@greatiam.com> Hello I am experimenting with FS and would like to know how to connect two independent servers with user on one beinng able to call users on the other. Do I set each server to be the gateway of the corresponding one ? Pardon me if this has already benn dealt with. My search has drawn a blank Thanks. From jbarou at sqli.com Wed Dec 2 01:54:05 2009 From: jbarou at sqli.com (Jonathan Barou) Date: Wed, 2 Dec 2009 10:54:05 +0100 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> Message-ID: <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> Any ideas ? Thanks 2009/11/30 Jonathan Barou > My version is FreeSWITCH Version 1.0.trunk (15691M) > > http://jira.freeswitch.org/browse/FSBUILD-213 > > Thanks you. > > 2009/11/27 Anthony Minessale > > by latest do you mean SVN trunk? >> >> Can you issue the command "sofia profile internal siptrace on" before >> capturing your trace and post the results >> to http://pastebin.freeswitch.org or open a jira >> http://jira.freeswitch.org on the issue and attach the log after you >> create the issue ticket, don't include it in the mailing list. >> >> >> On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: >> >>> Hi everybody, >>> >>> I'm actually using the lastest version of Freeswitch, I have a problem. I >>> have a trunk SIP with my PABX. >>> >>> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >>> 2. one Alcatel IpTouch 4028 with number 987 >>> (on PABX) >>> 3. one Siemens Gigaset A580 IP with number >>> 8401 (on Freeswitch) >>> >>> >>> *The first test* is to say to the phone 2 to transfer all the call to >>> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >>> phone 3 is ringing and it's work. I have that in the log : >>> >>> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >>> early media >>> >>> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> >>> 10.33.69.246 port 32000 codec: 8 ms: 90 >>> >>> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer >>> [soft] 720 bytes per 90ms >>> >>> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >>> >>> v=0 >>> >>> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >>> >>> s=FreeSWITCH >>> >>> c=IN IP4 10.33.169.92 >>> >>> t=0 0 >>> >>> m=audio 23054 RTP/AVP 8 106 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:106 telephone-event/8000 >>> >>> a=fmtp:106 0-16 >>> >>> a=silenceSupp:off - - - - >>> >>> a=ptime:90 >>> >>> a=sendrecv >>> >>> >>> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >>> sofia/internal/368 at 10.33.69.246! >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>> 368 at 10.33.69.246 skipping state [early][183] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >>> sofia/internal/368 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>> Activation Success L16 at 8000hz 1 channel 90ms >>> >>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >>> Ringback Tone [%(2000,4000,440.0,480.0)] >>> >>> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >>> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>> >>> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >>> confirmed. >>> >>> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >>> Buffer at 1440 bytes vs 81 >>> >>> >>> >>> *The Second Tes*t is to say to the phone 1 to transfer all the call to >>> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >>> one time and after it hangup. I have that in the log : >>> >>> >>> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >>> early media >>> >>> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> >>> 10.33.69.144 port 32000 codec: 8 ms: 90 >>> >>> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer >>> [soft] 720 bytes per 90ms >>> >>> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >>> >>> v=0 >>> >>> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >>> >>> s=FreeSWITCH >>> >>> c=IN IP4 10.33.169.92 >>> >>> t=0 0 >>> >>> m=audio 27732 RTP/AVP 8 106 >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> a=rtpmap:106 telephone-event/8000 >>> >>> a=fmtp:106 0-16 >>> >>> a=silenceSupp:off - - - - >>> >>> a=ptime:90 >>> >>> a=sendrecv >>> >>> >>> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >>> sofia/internal/987 at 10.33.69.246! >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>> 987 at 10.33.69.246 skipping state [early][183] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>> Activation Success L16 at 8000hz 1 channel 90ms >>> >>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >>> Ringback Tone [%(2000,4000,440.0,480.0)] >>> >>> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >>> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >>> 987 at 10.33.69.246 entering state [terminated][487] >>> >>> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >>> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >>> sofia/internal/987 at 10.33.69.246 [KILL] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 thread >>> mismatch skipping state handler. >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >>> original codec. >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >>> [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >>> other leg >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>> sofia/internal/sip:8401 at 10.33.170.231:5060 hanging up, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >>> sofia/internal/sip:8401 at 10.33.170.231:5060 >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >>> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >>> sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING >>> -> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on external >>> entities >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 thread >>> mismatch skipping state handler. >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 48 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to >>> sleep >>> >>> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot >>> create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >>> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>> >>> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >>> Cause: ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >>> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/987 at 10.33.69.246) State HANGUP >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>> sofia/internal/987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/987 at 10.33.69.246) State REPORTING >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >>> ORIGINATOR_CANCEL >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> >>> CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>> sofia/internal/987 at 10.33.69.246 [BREAK] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >>> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session 47 >>> (sofia/internal/987 at 10.33.69.246) Ended >>> >>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/987 at 10.33.69.246) State DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>> 987 at 10.33.69.246 SOFIA DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>> sofia/internal/987 at 10.33.69.246 Standard DESTROY >>> >>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >>> >>> Finally when I tried to call the phone 3 with the phone 1 it's working, >>> and not when I want to call the phone 3 with the phone 2, like just before, >>> it's ringing just one time and hangup. >>> >>> >>> Thanks you. >>> >>> >>> Best Regards >>> >>> -- >>> John >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jonathan BAROU > Groupe SQLI - CRCI > > 0472405368 > jbarou at sqli.com > > 1, place Verrazzano > 69258 LYON CEDEX 09 > > -- Jonathan BAROU Groupe SQLI - CRCI 0472405368 jbarou at sqli.com 1, place Verrazzano 69258 LYON CEDEX 09 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/2109ac3f/attachment-0002.html From sharad at coraltele.com Wed Dec 2 02:02:43 2009 From: sharad at coraltele.com (sharad) Date: Wed, 2 Dec 2009 15:32:43 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 42, Issue 12 References: Message-ID: <000f01ca7336$97bb1950$0c04a8c0@compaq77db609e> Hello We also faced the similar issue. Actually it is caused bacause hold on music files are missing. either you save all the music files or configure your dialplan accordingly. Sharad , Coral Telecom, India ----- Original Message ----- From: To: Sent: Wednesday, December 02, 2009 3:24 PM Subject: FreeSWITCH-users Digest, Vol 42, Issue 12 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > > 1. Re: OSP Interop w/ Trans Nexus (Max Clark) > 2. Re: CLIP on FXS channels with mod_openzap (Fran?ois Legal) > 3. Bridging/Connecting Freeswitch servers (Otis) > 4. Re: Transfer Problem (Jonathan Barou) > -------------------------------------------------------------------------------- > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From devel at thom.fr.eu.org Wed Dec 2 02:57:12 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 11:57:12 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks Fran?ois -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0229b88f/attachment-0002.html From kond at nstel.ru Wed Dec 2 00:34:42 2009 From: kond at nstel.ru (Nikolay Kondratyev) Date: Wed, 2 Dec 2009 11:34:42 +0300 Subject: [Freeswitch-users] call barge in Message-ID: <20091202083449.271C2116CD@mail.nstel.ru> Hi all, I'm evaluating FS for our organization. I must fulfill the following requirements: 1. Call recording: All (or selected) calls to the secretary must be recorded. 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes "emergency" (for example, an extension with emergency prefix) call to one of these subscribers -> Secretary barges in the established call (conference). 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. Can anybody please advise if this is possible with FS? If yes, is it just a configuration task, or additional programming will be needed? Thanks in advance, Nikolay. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/125993b7/attachment-0002.html From frank at carmickle.com Wed Dec 2 05:45:27 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 08:45:27 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Message-ID: <20091202134526.GR31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > > > Hello, > > I created an extension in my dialplan so that when an incoming > call arrives, it rings a group of lines and then fallback to the voicemail > if no line is answered. > > I wanted then that when voicemail starts, the > calling party could dial some numbers to fetch the voicemail. I used > bind_meta_app for this. My problem is, when using bind_meta_app, the > voicemail continues, and I sometimes experience freeswitch hanging after > the call is over, depending on when the bind_meta_app is activated. The action your looking for is what is bound to "*" in the default voicemail config. Look at autoload_configs/voicemail.conf.xml HTH --FC From frank at carmickle.com Wed Dec 2 06:01:55 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 09:01:55 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B162271.1010306@greatiam.com> References: <4B162271.1010306@greatiam.com> Message-ID: <20091202140155.GS31924@base.carmickle.com> On Wed, Dec 02, Otis wrote: > Hello > > I am experimenting with FS and would like to know how to connect two > independent servers with user on one beinng able to call users on the > other. Do I set each server to be the gateway of the corresponding one ? You can if you need them to authenticate to eachother. You have to decide on what you need. Do you not want extensions reachable from the public context? If not then you can do what I do. You can certainly put an ipv4 address in instead of the mangled ipv6 that's in this example. Then create an extension that matches on the extensions on the other machine and bridge them to the correct hostname and port. If you just want all the extensions reachable from the public context then do something like this in your dialplan/public.xml There are yet other ways to get this done. HTH --FC From erandr-junk at usa.net Wed Dec 2 06:47:47 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 06:47:47 -0800 (PST) Subject: [Freeswitch-users] Best way to run originate calls through dial plan Message-ID: <26610094.post@talk.nabble.com> What would be the best way of making originate() run call through a dial plan (compared to directly going to a specified VOIP gateway). Would it be loopbacks, i.e. smth like this? /opt/freeswitch/bin/fs_cli -x "originate {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js zzzzzzzzzz)'" The idea of this is that originate() sets up the first call, then webcall.js plays back a WAV, and bridges the first call with the second one (also set up via loopback). Thanks! -- View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26610094.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Wed Dec 2 07:23:53 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 2 Dec 2009 17:23:53 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name Message-ID: Hello, We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On the PRI there is a Nortel with Q.Sig. After a lot of configuration trials I've managed to set it to send back the connected name over the SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the connected name and then the Cisco adds it as a Remote-Party-ID). However, I did not save it and a power outage cleared this config. In my age I don't remember what I've done... Anyone knows the correct config? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/bb473cd7/attachment-0002.html From devel at thom.fr.eu.org Wed Dec 2 07:28:29 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 16:28:29 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202134526.GR31924@base.carmickle.com> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> Message-ID: <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> On Wed, 2 Dec 2009 08:45:27 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> >> >> Hello, >> >> I created an extension in my dialplan so that when an incoming >> call arrives, it rings a group of lines and then fallback to the >> voicemail >> if no line is answered. >> >> I wanted then that when voicemail starts, the >> calling party could dial some numbers to fetch the voicemail. I used >> bind_meta_app for this. My problem is, when using bind_meta_app, the >> voicemail continues, and I sometimes experience freeswitch hanging after >> the call is over, depending on when the bind_meta_app is activated. > > The action your looking for is what is bound to "*" in the default > voicemail config. Look at autoload_configs/voicemail.conf.xml > > > > HTH > --FC > I tried to remove the bind_meta_app from the dialplan, call the extension then press * when the greeting message starts, but it did not bring the voicemail prompt for my id and password. Fran?ois > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at freeswitch.org Wed Dec 2 07:46:04 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 2 Dec 2009 13:46:04 -0200 Subject: [Freeswitch-users] CDR records In-Reply-To: <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> Message-ID: What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions. JM On Tue, Dec 1, 2009 at 3:31 PM, Michael Collins wrote: > > > On Sun, Nov 29, 2009 at 10:06 AM, Pusk?s Zsolt wrote: > >> Hi Guys! >> >> I'm using the latest svn (15711) with the default xml config. Only >> modified >> cdr_csv.conf.xml the line to > name="legs" >> value="ab"/> >> >> Here is what i do: >> >> 1. 1000 calls 1001 (1001 answers the call) >> 2. 1001 do blind transfer to 1002 (using *1) >> 3. 1001 hangs up >> 4. 1002 answers the call >> 5. 1002 and 1000 hangs up >> >> 3 cdr records are generated (simplified): >> >> from,to,start,duration >> "1000" "1001" "2009-11-29 15:21:53" "53" "50" >> "1000" "1002" "2009-11-29 15:21:53" "79" "76" >> "1000" "1002" "2009-11-29 15:22:46" "26" "23" >> >> As you can see the second cdr is incorrect because 1000 doesn't speak with >> 1002 for 76 second. >> >> Is this a normal ? Is it possible to make only 2 record ? >> >> You may want to turn on mod_xml_curl and look at XML CDRs, comparing them > to the corresponding CSV files. That should help you figure out why the > values in the CSV are what they are. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/fbee6871/attachment-0002.html From frank at carmickle.com Wed Dec 2 08:00:11 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 11:00:11 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> Message-ID: <20091202160011.GU31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: Snip... > > voicemail config. Look at autoload_configs/voicemail.conf.xml > > > > > > > > HTH > > --FC > > > > I tried to remove the bind_meta_app from the dialplan, call the extension > then press * when the greeting message starts, but it did not bring the > voicemail prompt for my id and password. Did you check your voicemail config as I pointed out? autoload_configs/voicemail.conf.xml should have And what exactly do you mean by "Remote fetching of voicemail?" --FC From kristian.kielhofner at gmail.com Wed Dec 2 08:35:40 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 2 Dec 2009 11:35:40 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> Message-ID: <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> As always, you are correct. The scenario now is: - If the caller places the callee on hold, the callee will get hold music - If the callee places the caller on hold, the caller will not get hold music I've uploaded a fresh pastebin here: http://pastebin.freeswitch.org/11356 On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale wrote: > results cant possibly be the same > there is not even any broadcast involved in uuid_transfer ? > > you need to attach a console trace with debug log up > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From john_platts at hotmail.com Wed Dec 2 08:39:24 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 2 Dec 2009 10:39:24 -0600 Subject: [Freeswitch-users] Update to MODENDP-272 Message-ID: I have uploaded the dialplan and JavaScript files used to process calls to MODENDP-272. I have even done a make current to revision 15755, and the blind transfer is still failing. _________________________________________________________________ Windows 7: Unclutter your desktop. Learn more. http://www.microsoft.com/windows/windows-7/videos-tours.aspx?h=7sec&slideid=1&media=aero-shake-7second&listid=1&stop=1&ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_7secdemo:122009 From devel at thom.fr.eu.org Wed Dec 2 08:49:46 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 17:49:46 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202160011.GU31924@base.carmickle.com> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> <20091202160011.GU31924@base.carmickle.com> Message-ID: <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> I did check (and modify as my voicemail extension name is not vmain) the voicemail.conf.xml, and vmain-key is *. What I mean by remote fetching of voicemail, is that someone may dial in, either from inside (via FXS or even SIP) or outside (via FXO), then when reaching the voice mail to leave a message, he could dial some specific digit (or digits) to reach the voicemail login and fetch his voice mails. I can do this using bind_meta_app (it is already working), but then I need to terminate the extension when invoking the meta_app, otherwise freeswitch may sometimes hang if the meta app is called after the "leave a message" voicemail tone. Fran?ois On Wed, 2 Dec 2009 11:00:11 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: > Snip... >> > voicemail config. Look at autoload_configs/voicemail.conf.xml >> > >> > >> > >> > HTH >> > --FC >> > >> >> I tried to remove the bind_meta_app from the dialplan, call the extension >> then press * when the greeting message starts, but it did not bring the >> voicemail prompt for my id and password. > > Did you check your voicemail config as I pointed out? > autoload_configs/voicemail.conf.xml should have > > > > And what exactly do you mean by "Remote fetching of voicemail?" > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From frank at carmickle.com Wed Dec 2 09:16:14 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 12:16:14 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> <20091202134526.GR31924@base.carmickle.com> <0531eed0c9944866df58e2213cc1d0d9@thom.fr.eu.org> <20091202160011.GU31924@base.carmickle.com> <16015ed7cc1d66371f298c2e65b31eb2@thom.fr.eu.org> Message-ID: <20091202171613.GV31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > I did check (and modify as my voicemail extension name is not vmain) the > voicemail.conf.xml, and vmain-key is *. > > What I mean by remote fetching of voicemail, is that someone may dial in, > either from inside (via FXS or even SIP) or outside (via FXO), then when > reaching the voice mail to leave a message, he could dial some specific > digit (or digits) to reach the voicemail login and fetch his voice mails. > > I can do this using bind_meta_app (it is already working), but then I need > to terminate the extension when invoking the meta_app, otherwise freeswitch > may sometimes hang if the meta app is called after the "leave a message" > voicemail tone. Alright. I missed what vmain actually does in the voicemail config. It actually calls the extension named vmain in the dialplan. So if you don't have this then you will need to have one. Thanks for asking this question because my voicemail auth was broken and I didn't even know it! I fixed it and a working extension for vmain can look like this. HTH --FC From shiyanov at gmail.com Wed Dec 2 09:21:53 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 2 Dec 2009 20:21:53 +0300 Subject: [Freeswitch-users] call barge in In-Reply-To: <20091202083449.271C2116CD@mail.nstel.ru> References: <20091202083449.271C2116CD@mail.nstel.ru> Message-ID: 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Here is a bunch of code, search there ''barge" Artem On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote: > Hi all, > > > > I?m evaluating FS for our organization. > > I must fulfill the following requirements: > > 1. Call recording: All (or selected) calls to the secretary must be > recorded. > > 2. Call barge in: Assume that two subscribers are talking to each other. > Secretary makes ?emergency? (for example, an extension with emergency > prefix) call to one of these subscribers -> Secretary barges in the > established call (conference). > > 3. Call drop when emergency call arrives: the same as above, but > established call is dropped end emergency call is established. > > > > Can anybody please advise if this is possible with FS? > > If yes, is it just a configuration task, or additional programming will be > needed? > > > > Thanks in advance, > > Nikolay. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/c5807777/attachment-0002.html From abeka at greatiam.com Wed Dec 2 09:29:26 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 17:29:26 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202140155.GS31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> Message-ID: <4B16A3F6.6000702@greatiam.com> Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > >> Hello >> >> I am experimenting with FS and would like to know how to connect two >> independent servers with user on one beinng able to call users on the >> other. Do I set each server to be the gateway of the corresponding one ? >> > > You can if you need them to authenticate to eachother. You have to decide on what you need. Do you not want extensions reachable from the public context? If not then you can do what I do. > > > > > > > > > You can certainly put an ipv4 address in instead of the mangled ipv6 that's in this example. > > Then create an extension that matches on the extensions on the other machine and bridge them to the correct hostname and port. > > If you just want all the extensions reachable from the public context then do something like this in your dialplan/public.xml > > > > > > > > There are yet other ways to get this done. > > HTH > --FC > > > Thanks. I would like all extensions on say server A to be contactable by those on server B and vice versa. From msc at freeswitch.org Wed Dec 2 09:35:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:35:06 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912010931i7da0f743h7e023d75165e0bed@mail.gmail.com> Message-ID: <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> 2009/12/2 Jo?o Mesquita > What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions. > > JM > > Thanks for catching my typo! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/327f34ad/attachment-0002.html From devel at thom.fr.eu.org Wed Dec 2 09:43:32 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 02 Dec 2009 18:43:32 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: No, my voicemail extension (I have 2 actually) is called vmain_unregistered_user, so in voicemail.conf.xml I have : But still (and I don't even know if I'm using it the right way), I would expect that when the voicemail greeting starts, I could press * on the phone to call the vmain_unregistered_user extension in my dialplan, but that never happens. What I want, is when someone dials in (this is usefull only from outside so via FXO line) his extension (or any other, it's not important) that when he reach the voicemail to leave a message, override the "leave a message" voicemail and enter the "check my messages" voicemail (with an authentication step). I'm not sure this is clear. Fran?ois On Wed, 2 Dec 2009 12:16:14 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> I did check (and modify as my voicemail extension name is not vmain) the >> voicemail.conf.xml, and vmain-key is *. >> >> What I mean by remote fetching of voicemail, is that someone may dial in, >> either from inside (via FXS or even SIP) or outside (via FXO), then when >> reaching the voice mail to leave a message, he could dial some specific >> digit (or digits) to reach the voicemail login and fetch his voice mails. >> >> I can do this using bind_meta_app (it is already working), but then I >> need >> to terminate the extension when invoking the meta_app, otherwise >> freeswitch >> may sometimes hang if the meta app is called after the "leave a message" >> voicemail tone. > > Alright. I missed what vmain actually does in the voicemail config. It > actually calls the extension named vmain in the dialplan. So if you don't > have this then you will need to have one. Thanks for asking this question > because my voicemail auth was broken and I didn't even know it! I fixed it > and a working extension for vmain can look like this. > > > expression="^vmain$|^voicemail$|^\*98$|^\*86$"> > > > > > > > > HTH > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 2 09:47:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:47:49 -0800 Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <26610094.post@talk.nabble.com> References: <26610094.post@talk.nabble.com> Message-ID: <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: > > What would be the best way of making originate() run call through a dial > plan > (compared to directly going to a specified VOIP gateway). Would it be > loopbacks, i.e. smth like this? > > /opt/freeswitch/bin/fs_cli -x "originate > > {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML > '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js zzzzzzzzzz)'" > > The idea of this is that originate() sets up the first call, then > webcall.js > plays back a WAV, and bridges the first call with the second one (also set > up via loopback). > > Could you describe the problem that you're trying to solve? That would make it easier to know if what you've come up with is the best solution. How many calls per second were you wanting to generate with this setup? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/5b63db3d/attachment-0002.html From msc at freeswitch.org Wed Dec 2 09:50:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 09:50:03 -0800 Subject: [Freeswitch-users] Update to MODENDP-272 In-Reply-To: References: Message-ID: <87f2f3b90912020950g597b4217r6b198175044539f0@mail.gmail.com> On Wed, Dec 2, 2009 at 8:39 AM, John Platts wrote: > > I have uploaded the dialplan and JavaScript files used to process calls to > MODENDP-272. I have even done a make current to revision 15755, and the > blind transfer is still failing. > > John, Thanks for keeping the guys in the loop. Just a quick note: when you make updates to JIRA cases that you've opened the devs will see it so there's no need to send an email to the users list. :) Hold tight and the devs will check it out as soon as they can. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/4c89b033/attachment-0002.html From dlaperle at rsslex.com Wed Dec 2 07:00:25 2009 From: dlaperle at rsslex.com (David Laperle) Date: Wed, 02 Dec 2009 10:00:25 -0500 Subject: [Freeswitch-users] Dictation System Message-ID: <1259766025.1980.11.camel@dlaplap> Hi Freeswitch users, i'm new into the PBX world. I just installed FreeSwitch and made work great, but one of my goal with the PBX system is to use it as a dictation system. We were using Callweaver, and there's a Dictation module for CW and one for Asterisk, but i can't find one for FreeSwitch so far. Is there anything in the trunk that i could find or any work in progress? I'm willing to develop a bit to help the work in progress, i have the programming knowledge but not the VOIP/PBX knowledge, so a work in progress could be enough for me to start with and complete the work! If any of you have an idea or a hint for me i would be very grateful! Thanks a lot, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. Bien que ce courriel ait ?t? trait? contre les virus, il est de la responsabilit? du destinataire de s'assurer que l'envoi en est exempt. 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Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/de53ad84/attachment-0002.html From frank at carmickle.com Wed Dec 2 09:58:54 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 12:58:54 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B16A3F6.6000702@greatiam.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> Message-ID: <20091202175853.GW31924@base.carmickle.com> On Wed, Dec 02, Otis wrote: Snip... > Thanks. > > I would like all extensions on say server A to be contactable by those > on server B and vice versa. The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like http://wiki.freeswitch.org/wiki/Dialplan_XML http://wiki.freeswitch.org/wiki/Mod_dptools#Applications http://wiki.freeswitch.org/wiki/Sofia --FC From mctch at yahoo.com Wed Dec 2 09:58:52 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 2 Dec 2009 09:58:52 -0800 (PST) Subject: [Freeswitch-users] call barge in In-Reply-To: Message-ID: <619084.32069.qm@web56408.mail.re3.yahoo.com> 1. Call recording: All (or selected) calls to the secretary must be recorded. Just requires an addition to the dialplan.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes ?emergency? (for example, an extension with emergency prefix) call to one of thesesubscribers -> Secretary barges in the established call (conference). In FreeSWITCH it is called eavesdrop and its in the default configuration in the dialplan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdropTo use it from the default entry in the dialplan you can dial 88 followed by the extension number. If you want to limit it to a set of extensions just add an additional condition. 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. It might be possible to do it with just the xml dialplan. However it could definitely be accomplished within a short time using a dialplan entry and a lua, perl or javascript. Best Regards, Mark J Crane http://www.fusionpbx.com (Open source graphical interface for FreeSWITCH) --- On Wed, 12/2/09, Artem Shiyanov wrote: From: Artem Shiyanov Subject: Re: [Freeswitch-users] call barge in To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 2, 2009, 10:21 AM 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Here is a bunch of code, search there ''barge" Artem On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev wrote: Hi all, ? I?m evaluating FS for our organization. I must fulfill the following requirements: 1. Call recording: All (or selected) calls to the secretary must be recorded. 2. Call barge in: Assume that two subscribers are talking to each other. Secretary makes ?emergency? (for example, an extension with emergency prefix) call to one of these subscribers -> Secretary barges in the established call (conference). 3. Call drop when emergency call arrives: the same as above, but established call is dropped end emergency call is established. ? Can anybody please advise if this is possible with FS? If yes, is it just a configuration task, or additional programming will be needed? ? Thanks in advance, Nikolay. ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/70d2ee0a/attachment-0002.html From msc at freeswitch.org Wed Dec 2 10:02:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:02:26 -0800 Subject: [Freeswitch-users] call barge in In-Reply-To: References: <20091202083449.271C2116CD@mail.nstel.ru> Message-ID: <87f2f3b90912021002p3db48213l9dc666774fa9ab25@mail.gmail.com> On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov wrote: > 1 - config > 2 - I've done this with programming > 3 - suppose programming would be needed > > Just to clarify, when you say "programming" there are different levels of involvement. For example, you can do programming in C which is pretty in depth, but that's probably not what is required. Most likely this all can be done with dialplan configuration and some simple Lua/Perl/JavaScript scripts. (We support many scripting languages.) I recommend that you install FreeSWITCH on a test server and connect a few phones. Start with the default configuration and make sure that you have it working properly and go from there. Also, we have an IRC channel on irc.freenode.net where you can come and discuss things realtime. Lastly, we have a weekly conference call where you can ask community members and developers your questions: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call I recommend that you use the latest SVN trunk as we are really close to 1.0.5. If you're on a Linux box you can do the quick install process mentioned here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Dive in and have fun! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/a919b7fd/attachment-0002.html From frank at carmickle.com Wed Dec 2 10:05:48 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 13:05:48 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <20091202180548.GX31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : > > > > But still (and I don't even know if I'm using it the right way), I would > expect that when the voicemail greeting starts, I could press * on the > phone to call the vmain_unregistered_user extension in my dialplan, but > that never happens. > What I want, is when someone dials in (this is usefull only from outside > so via FXO line) his extension (or any other, it's not important) that > when > he reach the voicemail to leave a message, override the "leave a message" > voicemail and enter the "check my messages" voicemail (with an > authentication step). > > I'm not sure this is clear. Totally clear. My last example of extension vmain should have the goodies you need to have it work correctly. I have not tested changing to something different. Maybe this is broken? Try setting it back to the default and making your changes in the dialplan like I did. I can tell you that it is working as of svn 15396. --FC From msc at freeswitch.org Wed Dec 2 10:08:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:08:00 -0800 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <87f2f3b90912021008k1502b33dm434a6b19bf1f8792@mail.gmail.com> On Wed, Dec 2, 2009 at 9:43 AM, Fran?ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : > > > > But still (and I don't even know if I'm using it the right way), I would > expect that when the voicemail greeting starts, I could press * on the > phone to call the vmain_unregistered_user extension in my dialplan, but > that never happens. > What I want, is when someone dials in (this is usefull only from outside > so via FXO line) his extension (or any other, it's not important) that > when > he reach the voicemail to leave a message, override the "leave a message" > voicemail and enter the "check my messages" voicemail (with an > authentication step). > > I think you want to press zero: Of course, you can change that value to something else if you prefer the zero to be a transfer-to-operator kind of function. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/aa2211db/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 10:13:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:13:55 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> Message-ID: <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> I decided to just change the code so its more elegant to handle recursive broadcasting so you can try again and see if that helps. On Wed, Dec 2, 2009 at 10:35 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > As always, you are correct. > > The scenario now is: > > - If the caller places the callee on hold, the callee will get hold music > - If the callee places the caller on hold, the caller will not get hold > music > > I've uploaded a fresh pastebin here: > > http://pastebin.freeswitch.org/11356 > > On Fri, Nov 20, 2009 at 10:34 PM, Anthony Minessale > wrote: > > results cant possibly be the same > > there is not even any broadcast involved in uuid_transfer ? > > > > you need to attach a console trace with debug log up > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/545cd94a/attachment-0002.html From msc at freeswitch.org Wed Dec 2 10:13:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:13:59 -0800 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202175853.GW31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> Message-ID: <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > Snip... > > > Thanks. > > > > I would like all extensions on say server A to be contactable by those > > on server B and vice versa. > > The example I gave you should get you started. Let us know how you get > along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > Remember, too, that gateways are useful for doing auth/reg so having a gateway on each box that registers to the other box is pretty handy. If you run into any trouble trying to set it up you can ask here or join us in #freeswitch on irc.freenode.net. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/2b7e6c32/attachment-0002.html From frank at carmickle.com Wed Dec 2 10:15:28 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 2 Dec 2009 13:15:28 -0500 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: References: Message-ID: <20091202181527.GY31924@base.carmickle.com> On Wed, Dec 02, Fran??ois Legal wrote: > No, my voicemail extension (I have 2 actually) is called > vmain_unregistered_user, so in voicemail.conf.xml I have : Also, is there a functional requirement for two different extensions to call vmain? --FC From anthony.minessale at gmail.com Wed Dec 2 10:18:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:18:48 -0600 Subject: [Freeswitch-users] Dictation System In-Reply-To: <1259766025.1980.11.camel@dlaplap> References: <1259766025.1980.11.camel@dlaplap> Message-ID: <191c3a030912021018h78776ec0w51f18df795b78c1b@mail.gmail.com> Yes, I'm familiar with that application, check the src code for the author =p There has not been much of a demand for such an application but it's of course entirely possible to develop one. On Wed, Dec 2, 2009 at 9:00 AM, David Laperle wrote: > Hi Freeswitch users, > > i'm new into the PBX world. I just installed FreeSwitch and made work > great, but one of my goal with the PBX system is to use it as a dictation > system. We were using Callweaver, and there's a Dictation module for CW and > one for Asterisk, but i can't find one for FreeSwitch so far. Is there > anything in the trunk that i could find or any work in progress? > > I'm willing to develop a bit to help the work in progress, i have the > programming knowledge but not the VOIP/PBX knowledge, so a work in progress > could be enough for me to start with and complete the work! > > If any of you have an idea or a hint for me i would be very grateful! > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > *Robinson Sheppard Shapiro *s.e.n.c.r.l/LLP > Avocats / Barristers & Solicitors > 4600 - 800 Place Victoria > Montr?al Qc H4Z 1H6 > T (514) 878-2631 F (514) 878-1865 > www.rsslex.com et/and www.rsscanadaimmigration.com > > > > > * > ------------------------------ > **http://www.rsslex.com** * > > *AVIS:* Ce courriel privil?gi? et confidentiel est destin? ? la seule > personne ou entit? ? laquelle il est adress?. Pour toute autre personne, > toute action prise en rapport ? ce courriel ainsi que toute lecture, > reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble > de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? > recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. > Bien que ce courriel ait ?t? trait? contre les virus, il est de la > responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos > communications avec vous peuvent contenir des renseignements confidentiels > ou prot?g?s par le secret professionnel. Si vous d?sirez que nous > communiquions avec vous par un autre moyen de transmission que le courrier > ?lectronique ordinaire non s?curis?, veuillez nous en aviser. > > *NOTICE:* This privileged and confidential email is intended only for the > individual or entity to whom it is addressed. With regard to all others, any > action related with this email as well as any reading, reproduction, > transmission and/or dissemination in whole or in part of the information > included in this email is prohibited. If you are not the addressee, > immediately return the email to sender prior to destroying all copies. Even > if this email is believed to be free from any virus, it is the > responsibility of the recipient to make sure that it is virus exempt. Our > communications to you may contain confidential information or information > protected under solicitor-client privilege. Please advise if you wish us to > use a mode of communication other than regular, unsecured e-mail in our > communications with you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/7f0aaee3/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 10:21:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:21:54 -0600 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> References: <5f0c783ed3de925ba34a8c12170359f6@thom.fr.eu.org> Message-ID: <191c3a030912021021n23fa5981q8b6398587dfcb739@mail.gmail.com> bind to the transfer app so that it transfers the call to the vm extension that way the current application is always interrupted and replaced. The special "inline" dialplan lets you transfer calls right to an application use "inline" as the dp name and voicemail: as the extension On Wed, Dec 2, 2009 at 4:57 AM, Fran?ois Legal wrote: > Hello, > > > > I created an extension in my dialplan so that when an incoming call > arrives, it rings a group of lines and then fallback to the voicemail if no > line is answered. > > I wanted then that when voicemail starts, the calling party could dial some > numbers to fetch the voicemail. I used bind_meta_app for this. My problem > is, when using bind_meta_app, the voicemail continues, and I sometimes > experience freeswitch hanging after the call is over, depending on when the > bind_meta_app is activated. > > How can I make freeswitch terminate the first voicemail instance when > activating the bind_meta_app. > > > > Here's my extension : > > > > > > > > > > > > > > > Thanks > > > > Fran?ois > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0c04d795/attachment-0002.html From msc at freeswitch.org Wed Dec 2 10:23:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 10:23:01 -0800 Subject: [Freeswitch-users] Dictation System In-Reply-To: <1259766025.1980.11.camel@dlaplap> References: <1259766025.1980.11.camel@dlaplap> Message-ID: <87f2f3b90912021023x138c3365t908ccd9cd877f66b@mail.gmail.com> This seems like an interesting niche project. I think that if you have programming skills then the community can provide the PBX/VoIP knowledge to help you get over the hump. I would recommend that you write up a document describing all the features that this module would need to provide. Reply to this thread when you have that ready. In the meantime I'll ask the community now: is there anyone else interested in this functionality that would like to help David get it off the ground? David, if you haven't already done so I recommend joining the freeswitch-dev mailing list and hopping on IRC in #freeswitch and #freeswitch-dev over on irc.freenode.net. Don't forget that we also have a weekly FreeSWITCH conference call each Friday so you can always hop on and discuss your ideas with some of the devs and other community members: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call Welcome to the community! -Michael (IRC: mercutioviz) On Wed, Dec 2, 2009 at 7:00 AM, David Laperle wrote: > Hi Freeswitch users, > > i'm new into the PBX world. I just installed FreeSwitch and made work > great, but one of my goal with the PBX system is to use it as a dictation > system. We were using Callweaver, and there's a Dictation module for CW and > one for Asterisk, but i can't find one for FreeSwitch so far. Is there > anything in the trunk that i could find or any work in progress? > > I'm willing to develop a bit to help the work in progress, i have the > programming knowledge but not the VOIP/PBX knowledge, so a work in progress > could be enough for me to start with and complete the work! > > If any of you have an idea or a hint for me i would be very grateful! > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/50c00df2/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 10:23:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:23:11 -0600 Subject: [Freeswitch-users] Transfer Problem In-Reply-To: <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> References: <8048ff7f0911270847h2c270cact51ca9a51017db12d@mail.gmail.com> <191c3a030911270903i341d1f83pa15f67443422cb67@mail.gmail.com> <8048ff7f0911300033u45c7aa5cwca16581ef9a22c2b@mail.gmail.com> <8048ff7f0912020154p20f962a5i3f954e08d1d5fd2d@mail.gmail.com> Message-ID: <191c3a030912021023y3663b48dgab739ad2e7cb95c@mail.gmail.com> you may have nat problems since I do not see many people reporting anything like this. On Wed, Dec 2, 2009 at 3:54 AM, Jonathan Barou wrote: > Any ideas ? > > Thanks > > 2009/11/30 Jonathan Barou > > My version is FreeSWITCH Version 1.0.trunk (15691M) >> >> http://jira.freeswitch.org/browse/FSBUILD-213 >> >> Thanks you. >> >> 2009/11/27 Anthony Minessale >> >> by latest do you mean SVN trunk? >>> >>> Can you issue the command "sofia profile internal siptrace on" before >>> capturing your trace and post the results >>> to http://pastebin.freeswitch.org or open a jira >>> http://jira.freeswitch.org on the issue and attach the log after you >>> create the issue ticket, don't include it in the mailing list. >>> >>> >>> On Fri, Nov 27, 2009 at 10:47 AM, Jonathan Barou wrote: >>> >>>> Hi everybody, >>>> >>>> I'm actually using the lastest version of Freeswitch, I have a problem. >>>> I have a trunk SIP with my PABX. >>>> >>>> There is 3 phones : 1. one Alcatel Advanced with number 368 (on PABX) >>>> 2. one Alcatel IpTouch 4028 with number 987 >>>> (on PABX) >>>> 3. one Siemens Gigaset A580 IP with number >>>> 8401 (on Freeswitch) >>>> >>>> >>>> *The first test* is to say to the phone 2 to transfer all the call to >>>> number 8401. So when I dial 987 on the phone 1, all work perfectly, the >>>> phone 3 is ringing and it's work. I have that in the log : >>>> >>>> 2009-11-27 16:52:18.677299 [INFO] switch_ivr_originate.c:1024 Sending >>>> early media >>>> >>>> 2009-11-27 16:52:18.677299 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>>> [sofia/internal/368 at 10.33.69.246] 10.33.169.92 port 23054 -> >>>> 10.33.69.246 port 32000 codec: 8 ms: 90 >>>> >>>> 2009-11-27 16:52:18.677299 [DEBUG] switch_rtp.c:1155 Starting timer >>>> [soft] 720 bytes per 90ms >>>> >>>> 2009-11-27 16:52:18.687301 [INFO] mod_sofia.c:1706 Ring SDP: >>>> >>>> v=0 >>>> >>>> o=FreeSWITCH 1259314084 1259314085 IN IP4 10.33.169.92 >>>> >>>> s=FreeSWITCH >>>> >>>> c=IN IP4 10.33.169.92 >>>> >>>> t=0 0 >>>> >>>> m=audio 23054 RTP/AVP 8 106 >>>> >>>> a=rtpmap:8 PCMA/8000 >>>> >>>> a=rtpmap:106 telephone-event/8000 >>>> >>>> a=fmtp:106 0-16 >>>> >>>> a=silenceSupp:off - - - - >>>> >>>> a=ptime:90 >>>> >>>> a=sendrecv >>>> >>>> >>>> 2009-11-27 16:52:18.687301 [NOTICE] mod_sofia.c:1709 Pre-Answer >>>> sofia/internal/368 at 10.33.69.246! >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:706 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:412 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>>> 368 at 10.33.69.246 skipping state [early][183] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_core_session.c:645 Send signal >>>> sofia/internal/368 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>>> Activation Success L16 at 8000hz 1 channel 90ms >>>> >>>> 2009-11-27 16:52:18.687301 [DEBUG] switch_ivr_originate.c:1116 Play >>>> Ringback Tone [%(2000,4000,440.0,480.0)] >>>> >>>> 2009-11-27 16:52:18.747333 [DEBUG] switch_core_io.c:652 sofia/internal/ >>>> 368 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>>> >>>> 2009-11-27 16:52:18.927433 [DEBUG] switch_rtp.c:1992 Correct ip/port >>>> confirmed. >>>> >>>> 2009-11-27 16:52:19.187876 [DEBUG] switch_core_io.c:402 Engaging Read >>>> Buffer at 1440 bytes vs 81 >>>> >>>> >>>> >>>> *The Second Tes*t is to say to the phone 1 to transfer all the call to >>>> number 8401. So when I dial 368 on the phone 2, the phone 3 is ringing just >>>> one time and after it hangup. I have that in the log : >>>> >>>> >>>> 2009-11-27 17:17:10.487610 [INFO] switch_ivr_originate.c:1024 Sending >>>> early media >>>> >>>> 2009-11-27 17:17:10.487610 [DEBUG] sofia_glue.c:2375 AUDIO RTP >>>> [sofia/internal/987 at 10.33.69.246] 10.33.169.92 port 27732 -> >>>> 10.33.69.144 port 32000 codec: 8 ms: 90 >>>> >>>> 2009-11-27 17:17:10.487610 [DEBUG] switch_rtp.c:1155 Starting timer >>>> [soft] 720 bytes per 90ms >>>> >>>> 2009-11-27 17:17:10.497659 [INFO] mod_sofia.c:1706 Ring SDP: >>>> >>>> v=0 >>>> >>>> o=FreeSWITCH 1259310898 1259310899 IN IP4 10.33.169.92 >>>> >>>> s=FreeSWITCH >>>> >>>> c=IN IP4 10.33.169.92 >>>> >>>> t=0 0 >>>> >>>> m=audio 27732 RTP/AVP 8 106 >>>> >>>> a=rtpmap:8 PCMA/8000 >>>> >>>> a=rtpmap:106 telephone-event/8000 >>>> >>>> a=fmtp:106 0-16 >>>> >>>> a=silenceSupp:off - - - - >>>> >>>> a=ptime:90 >>>> >>>> a=sendrecv >>>> >>>> >>>> 2009-11-27 17:17:10.497659 [NOTICE] mod_sofia.c:1709 Pre-Answer >>>> sofia/internal/987 at 10.33.69.246! >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:706 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:412 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 receive message [DISPLAY] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] sofia.c:3691 Channel sofia/internal/ >>>> 987 at 10.33.69.246 skipping state [early][183] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_core_session.c:645 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1054 Raw Codec >>>> Activation Success L16 at 8000hz 1 channel 90ms >>>> >>>> 2009-11-27 17:17:10.497659 [DEBUG] switch_ivr_originate.c:1116 Play >>>> Ringback Tone [%(2000,4000,440.0,480.0)] >>>> >>>> 2009-11-27 17:17:10.537273 [DEBUG] switch_core_io.c:652 sofia/internal/ >>>> 987 at 10.33.69.246 receive message [TRANSCODING_NECESSARY] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] sofia.c:3696 Channel sofia/internal/ >>>> 987 at 10.33.69.246 entering state [terminated][487] >>>> >>>> 2009-11-27 17:17:11.317096 [NOTICE] sofia.c:4299 Hangup sofia/internal/ >>>> 987 at 10.33.69.246 [CS_EXECUTE] [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_channel.c:1912 Send signal >>>> sofia/internal/987 at 10.33.69.246 [KILL] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.317096 [DEBUG] switch_core_state_machine.c:459 >>>> thread mismatch skipping state handler. >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_codec.c:122 Restore >>>> original codec. >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_ivr_originate.c:2842 Hangup >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_CONSUME_MEDIA] >>>> [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_channel.c:1912 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [KILL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 Overriding SIP cause 487 with 487 from the >>>> other leg >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 hanging up, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:406 Sending CANCEL to >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard HANGUP, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State HANGUP going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard REPORTING, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State REPORTING going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State Change CS_REPORTING >>>> -> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 48 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Locked, Waiting on >>>> external entities >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:459 >>>> thread mismatch skipping state handler. >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2982 Originate >>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session >>>> 48 (sofia/internal/sip:8401 at 10.33.170.231:5060) Ended >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>>> Channel sofia/internal/sip:8401 at 10.33.170.231:5060 [CS_DESTROY] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) Running State Change >>>> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>>> sip:8401 at 10.33.170.231:5060 SOFIA DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/internal/sip:8401 at 10.33.170.231:5060 Standard DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/sip:8401 at 10.33.170.231:5060) State DESTROY going to >>>> sleep >>>> >>>> 2009-11-27 17:17:11.347287 [ERR] switch_ivr_originate.c:2248 Cannot >>>> create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_ivr_originate.c:2988 Originate >>>> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >>>> >>>> 2009-11-27 17:17:11.347287 [INFO] mod_dptools.c:2295 Originate Failed. >>>> Cause: ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:348 >>>> (sofia/internal/987 at 10.33.69.246) State EXECUTE going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/987 at 10.33.69.246) State HANGUP >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:352 sofia/internal/ >>>> 987 at 10.33.69.246 Overriding SIP cause 487 with 487 from the other leg >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:358 Channel >>>> sofia/internal/987 at 10.33.69.246 hanging up, cause: ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/internal/987 at 10.33.69.246 Standard HANGUP, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:486 >>>> (sofia/internal/987 at 10.33.69.246) State HANGUP going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:333 >>>> (sofia/internal/987 at 10.33.69.246) State Change CS_HANGUP -> >>>> CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:314 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/987 at 10.33.69.246) State REPORTING >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/internal/987 at 10.33.69.246 Standard REPORTING, cause: >>>> ORIGINATOR_CANCEL >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:577 >>>> (sofia/internal/987 at 10.33.69.246) State REPORTING going to sleep >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:327 >>>> (sofia/internal/987 at 10.33.69.246) State Change CS_REPORTING -> >>>> CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:984 Send signal >>>> sofia/internal/987 at 10.33.69.246 [BREAK] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_session.c:1121 Session 47 >>>> (sofia/internal/987 at 10.33.69.246) Locked, Waiting on external entities >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1139 Session >>>> 47 (sofia/internal/987 at 10.33.69.246) Ended >>>> >>>> 2009-11-27 17:17:11.347287 [NOTICE] switch_core_session.c:1141 Close >>>> Channel sofia/internal/987 at 10.33.69.246 [CS_DESTROY] >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:423 >>>> (sofia/internal/987 at 10.33.69.246) Running State Change CS_DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/987 at 10.33.69.246) State DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] mod_sofia.c:293 sofia/internal/ >>>> 987 at 10.33.69.246 SOFIA DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/internal/987 at 10.33.69.246 Standard DESTROY >>>> >>>> 2009-11-27 17:17:11.347287 [DEBUG] switch_core_state_machine.c:434 >>>> (sofia/internal/987 at 10.33.69.246) State DESTROY going to sleep >>>> >>>> Finally when I tried to call the phone 3 with the phone 1 it's working, >>>> and not when I want to call the phone 3 with the phone 2, like just before, >>>> it's ringing just one time and hangup. >>>> >>>> >>>> Thanks you. >>>> >>>> >>>> Best Regards >>>> >>>> -- >>>> John >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Jonathan BAROU >> Groupe SQLI - CRCI >> >> 0472405368 >> jbarou at sqli.com >> >> 1, place Verrazzano >> 69258 LYON CEDEX 09 >> >> > > > -- > Jonathan BAROU > Groupe SQLI - CRCI > > 0472405368 > jbarou at sqli.com > 1, place Verrazzano > 69258 LYON CEDEX 09 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0a39440f/attachment-0002.html From erandr-junk at usa.net Wed Dec 2 10:23:51 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 10:23:51 -0800 (PST) Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> References: <26610094.post@talk.nabble.com> <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> Message-ID: <26613841.post@talk.nabble.com> I need a way to start a call from the PHP script to the originating number, tell the party on that number to hold on, start another call to destination number, and bridge everything together. On both legs I need to pass custom caller ID. I can of course open direct connections to VOIP gateways right from PHP, but I want to reuse existing routing rules in the dial plan, hence I want to know what's the best way of making originate go through a specific context of the dial plan. As for the number of calls per second, it's going to be only occasionally used. mercutioviz wrote: > > On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: > >> >> What would be the best way of making originate() run call through a dial >> plan >> (compared to directly going to a specified VOIP gateway). Would it be >> loopbacks, i.e. smth like this? >> >> /opt/freeswitch/bin/fs_cli -x "originate >> >> {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML >> '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js >> zzzzzzzzzz)'" >> >> The idea of this is that originate() sets up the first call, then >> webcall.js >> plays back a WAV, and bridges the first call with the second one (also >> set >> up via loopback). >> >> > Could you describe the problem that you're trying to solve? That would > make > it easier to know if what you've come up with is the best solution. How > many > calls per second were you wanting to generate with this setup? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26613841.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Wed Dec 2 10:32:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 12:32:21 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> References: <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> Message-ID: <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> idle is a 4 letter word to a realtime application. The core keeps a single high-priority thread to keep 1ms timing and expands that broadcasting to hundreds or thousand of threads who need accurate timing. Your choppy audio is caused by linksys lying about the packet len that it's using and we set our timer to the wrong speed. On Tue, Dec 1, 2009 at 9:19 PM, wrote: > Wow... Thinking about this timer setting and about how it converted > send()/recv() from non-blocking to blocking, I straced freeswitch when it > was > supposed to be idle. It never pauses! It keeps going in and out of select() > every millisecond! Why?? > > ------ Original Message ------ > Received: Tue, 01 Dec 2009 08:31:46 PM EST > From: erandr-junk at usa.net > To: > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. > Just > > installing SVN trunk didn't fix it either, but setting that option > afterwards > > surely did the trick. > > > > One thing I've noticed while staring at the console is that it *looks > like* > > that w/o the new setting the stuttering happens when FS either > re-registers > > itself with the provider or one of the SPA's port re-registers with FS. > > > > ------ Original Message ------ > > Received: Tue, 01 Dec 2009 05:33:26 PM EST > > From: Anthony Minessale > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Choppy sound with PCMU > > > > > linksys has had a bug for eons that can be fixed by setting the ptime > (or > > > rtp packet size in their terms) > > > in it's firmware to .20 instead of .30 > > > > > > Asterisk does not use async RTP like we do so it's never a problem > > > you can disable the timer by setting the channel var > rtp_timer_name=none > or > > > sofia param rtp-timer-name to none in the sofia profile. > > > > > > You should also test this on latest SVN trunk or wait for pre8 > > > > > > > > > > > > On Tue, Dec 1, 2009 at 3:52 PM, eaf wrote: > > > > > > > > > > > I should also add, after browsing through some topics here, that my > SIP > > > > provider sends 172-byte RTP frames, which is in accordance with > ptime:20 > > > > that it gives to FreeSWITCH. > > > > > > > > > > > > eaf wrote: > > > > > > > > > > Hi, > > > > > > > > > > I'm trying to migrate from Asterisk to FreeSWITCH (really like the > way > > > > how > > > > > it can be programmed), but ran into one issue with sound quality > that > I > > > > > just cannot workaround by myself. I would describe the sound > problem > as > > > > > being "choppy". From time to time small portions of the other > party's > > > > > voice are dropped, so the voice kind of stutters. This is not too > bad, > > > > but > > > > > is really noticeable, happens in every call and I don't experience > the > > > > > same with Asterisk running on the same box. I attached two files: > > > > > freeswitch.wav and asterisk.mp3 to illustrate my point. > > > > > > > > > > Issue completely goes away, if I set inbound-proxy-media to true. > > > > > > > > > > The way how I test is to connect SPA-2000 via 10mbps LAN to the box > > > > > directly exposed to internet, and then dial a toll-free via > FutureNine > > (a > > > > > SIP provider). > > > > > > > > > > The codec in use is PCMU. Can't really try PCMA or anything else > with > > > > this > > > > > provider. Only PCMU. Tried to match ptime of provider (30) with > ptime > > of > > > > > the SPA, didn't get any improvement. Tried turning off recording, > no > > > > > change either. > > > > > > > > > > What puzzles me is that even with greedy codec negotiations and > with > > PCMU > > > > > on both sides of FreeSWITCH, it's still saying that > > > > > TRANSCODING_NECESSARY. I'm attaching relevant portion of > freeswitch.log > > > > to > > > > > illustrate. > > > > > > > > > > The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode > > LX800 > > > > > with 997 bogomips. 256MB RAM. Only one call in progress, so I hope > that > > > > > it's not a performance issue. > > > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.wavfreeswitch.wav > > > > > > > > > > http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 > > > > > > > > > > http://old.nabble.com/file/p26594250/freeswitch.logfreeswitch.log > > > > > > > > > > Tried both 1.0.4 and 1.0.5pre5. Same results. > > > > > > > > > > What should I do next? Calls are consistently bad with FreeSWITCH, > and > > > > > consistently show no glitches with Asterisk. > > > > > > > > > > > > > > > > > > -- > > > > View this message in context: > > > > > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26599565.html > > > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com< > MSN%3Aanthony_minessale at hotmail.com > > > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org< > sip%3A888 at conference.freeswitch.org > > > > > iax:guest at conference.freeswitch.org/888 > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/a320b89e/attachment-0002.html From davis.erwin at gmail.com Wed Dec 2 10:41:56 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 13:41:56 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match Message-ID: Hi, I got a weird issue when I dialed an extension and listen to a recorded voice mail greeting message. After playing a couple of time of the greeting, the FS printed the warning of "sample rate not matching", then send the audio to a different remote RTP port. See the log below, 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate doesn't match 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec Activated 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original codec. 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less than minimum record length: 3, discarding it. 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 16000hz 1 channels 20ms 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ the original codec is wideband 16kHz Speex and the wireshark shows that the FS used the same codec. I used FS 1.04 in fedora 8. I have two questions here, (1) why does FS report "Sample rate doesn't match"? is it a bug or configuration issue? (2) Why does FS change the RTP port ? how to fix it? Thanks, Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/8d7045db/attachment-0002.html From shiyanov at gmail.com Wed Dec 2 11:08:17 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 2 Dec 2009 22:08:17 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: I'm back again with the same issue. Now it is became worse: it reproduces occasionally. [FS version is 1.04, test_load = 2 active calls] I've got 2 logs: successful and not. Here is a bad_case: 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute java(/usr/local/freeswitch/scripts/fs2agi.jar org.starpound.fs2agi.Translator ${agi_url}) Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run INFO: *************************************************** Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI application agi://localhost:4573/hello.agi?callId=929 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/2001! 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel [sofia/internal/2001] has been answered Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: java.lang.Exception: Internal FreeSwitch failure while streamming file, see FreeSwitch logs for details at org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) at org.starpound.fs2agi.Translator.run(Translator.java:56) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.freeswitch.Launcher.launch(Launcher.java:80) 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for details. 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 (sofia/external/6786081291 at 66.19.38.143) Ended 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 (sofia/internal/2001) Ended 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2001 [CS_DESTROY] Message "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: ..." is sent from my app upon the onHangup().` And here is a good_case: 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute java(/usr/local/freeswitch/scripts/fs2agi.jar org.starpound.fs2agi.Translator ${agi_url}) Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run INFO: *************************************************** Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for session 7c37369b-ffb2-4436-9288-a640047d0e5e 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI application agi://localhost:4573/hello.agi?callId=932 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/2001! 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel [sofia/internal/2001] has been answered Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: java.lang.Exception: Internal FreeSwitch failure while streamming file, see FreeSwitch logs for details at org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) at org.starpound.fs2agi.Translator.run(Translator.java:56) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.freeswitch.Launcher.launch(Launcher.java:80) 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for details. 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from 172.26.10.39:26402 to 91.190.120.190:26402 Suggestions? On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: > Anthony, > > >>As soon as you call uuid_bridge you are transferring both legs of the > call to bridge to each other. > >>This means your java app must exit so the channels can connect to each > other. > > I didn't know that. Now my java app is exiting upon the onHangup() call so > everything has become "ok". Thank you much. > I'll add note to the wiki about this issue. > > Artem > > > > > On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Your "annoying behaviour" is the exact behavior you should be getting >> considering what you told FS to do. >> >> As soon as you call uuid_bridge you are transferring both legs of the call >> to bridge to each other. >> This means your java app must exit so the channels can connect to each >> other. >> >> remember that you hangup hook can be called when the channel is >> transferred not only when it hangs up. >> you have to test which is happening based on the input to your callback. >> >> >> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >> >>> Hi there! >>> >>> I've got annoying FS behavior: >>> There are 2 channels executing the same Java application (application >>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>> channels are killed. Here is a log from FS console: >>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>> CS_HIBERNATE >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>> CS_EXECUTE -> CS_HIBERNATE >>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook called >>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>> >>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>> switch_core_session.c:933 Send signal >>> sofia/internal/1001 at master.agent.starpoundtec >>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>> >>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >>> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>> >>> (FS version is 1.0.4) >>> >>> Any thoughts? >>> >>> >>> Artem >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/f08f1532/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 11:17:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:17:55 -0600 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> Message-ID: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: > So I did some tests and still I can not see CLIP on a phone connected to an > FXS port. Whether the call is bridged from SIP UA or from an incoming call > on FXO port does not change anything. Whether the parameter > enable-caller-id=true is present or not in openzap.conf.xml does not change > anything too. > > On that subject, sangoma support team says it must be freeswitch as this > feature is supported and has been tested working. > > > > However, the good point is that I did not experience cuts in my call > bridged from FXS to FXO with that new release. > > > > Fran?ois > > > > On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > upgrading always helps *something* not sure. but that is where we have to > start because we have changed that code alot. > > > On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: > >> Sure, I'll try that. I'm just building freeswitch-snapshot that I >> downloaded from files.freeswitch.org >> >> I also experience, when bridging a call from an FXS to FXO the call is cut >> after a random time (this does not appear when bridging SIP to FXO). Might >> this upgrade fix this problem also ? >> >> >> >> Fran?ois >> >> >> >> On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: >> >> can you test svn trunk or latest pre release of 1.0.5 >> >> >> On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: >> >>> Hello, >>> >>> >>> >>> I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP >>> problems on the FXS ports. >>> >>> When I ring on FXS ports, the connected phone does not display >>> callerid/callerid-name. >>> >>> I tried turning the stuff of in openzap.conf.xml () but it did not help. >>> >>> >>> >>> As a side note, turning this on on the FXO ports drops the callerid >>> information on incoming calls. >>> >>> >>> >>> Running freeswitch 1.0.4 on linux 2.6.27. >>> >>> >>> >>> Fran?ois >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/65c5d3d2/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 11:24:09 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:24:09 -0600 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> Message-ID: <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> you should be working on SVN trunk if you are doing development, we are so far forward from 1.0.4 we can't do debugging very easily. I don't know all of the details of what you are trying to do but you are hitting some race conditions because of the async nature of the socket connection and the way you are using it. On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov wrote: > I'm back again with the same issue. > Now it is became worse: it reproduces occasionally. > [FS version is 1.04, test_load = 2 active calls] > > I've got 2 logs: successful and not. > Here is a bad_case: > > 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute > java(/usr/local/freeswitch/scripts/fs2agi.jar > org.starpound.fs2agi.Translator > ${agi_url}) > Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run > INFO: *************************************************** > Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run > INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for > session > 2898ad41-4ec1-4628-89fd-651a93a7221d > 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI application > agi://localhost:4573/hello.agi?callId=929 > 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream > handle! > > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] > 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/2001! > 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel > [sofia/internal/2001] has > been answered > Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session > 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: > java.lang.Exception: Internal FreeSwitch failure while streamming file, see > FreeSwitch logs for details > at > > org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) > at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) > at org.starpound.fs2agi.Translator.run(Translator.java:56) > at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) > at > > sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) > at > > sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) > at java.lang.reflect.Method.invoke(Method.java:597) > at org.freeswitch.Launcher.launch(Launcher.java:80) > 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup > sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application > agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for > details. > 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup > sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/external/6786081291 at 66.19.38.143) Ended > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 > (sofia/internal/2001) Ended > 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/internal/2001 [CS_DESTROY] > > > > Message > "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for session > 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: > ..." > is sent from my app upon the onHangup().` > > And here is a good_case: > > 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute > java(/usr/local/freeswitch/scripts/fs2agi.jar > org.starpound.fs2agi.Translator > ${agi_url}) > Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run > INFO: *************************************************** > Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run > INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for > session > 7c37369b-ffb2-4436-9288-a640047d0e5e > 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI application > agi://localhost:4573/hello.agi?callId=932 > 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream > handle! > > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] > 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready > sofia/internal/2001! > 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel > [sofia/internal/2001] has > been answered > Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed > INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for session > 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: > java.lang.Exception: Internal FreeSwitch failure while streamming file, see > FreeSwitch logs for details > at > > org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) > at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) > at org.starpound.fs2agi.Translator.run(Translator.java:56) > at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) > at > > sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) > at > > sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) > at java.lang.reflect.Method.invoke(Method.java:597) > at org.freeswitch.Launcher.launch(Launcher.java:80) > 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application > agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for > details. > 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port from > 172.26.10.39:26402 to 91.190.120.190:26402 > > > > Suggestions? > > > > > > > > > > > > On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: > >> Anthony, >> >> >>As soon as you call uuid_bridge you are transferring both legs of the >> call to bridge to each other. >> >>This means your java app must exit so the channels can connect to each >> other. >> >> I didn't know that. Now my java app is exiting upon the onHangup() call so >> everything has become "ok". Thank you much. >> I'll add note to the wiki about this issue. >> >> Artem >> >> >> >> >> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Your "annoying behaviour" is the exact behavior you should be getting >>> considering what you told FS to do. >>> >>> As soon as you call uuid_bridge you are transferring both legs of the >>> call to bridge to each other. >>> This means your java app must exit so the channels can connect to each >>> other. >>> >>> remember that you hangup hook can be called when the channel is >>> transferred not only when it hangs up. >>> you have to test which is happening based on the input to your callback. >>> >>> >>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >>> >>>> Hi there! >>>> >>>> I've got annoying FS behavior: >>>> There are 2 channels executing the same Java application (application >>>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>>> channels are killed. Here is a log from FS console: >>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>>> CS_HIBERNATE >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>> called >>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>>> playing file >>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>>> playing file >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send signal >>>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>>> CS_EXECUTE -> CS_HIBERNATE >>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>> called >>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>> >>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>>> switch_core_session.c:933 Send signal >>>> sofia/internal/1001 at master.agent.starpoundtec >>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send signal >>>> sofia/internal/1005 at 192.168.147.130 [BREAK] >>>> >>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send signal >>>> sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>>> >>>> (FS version is 1.0.4) >>>> >>>> Any thoughts? >>>> >>>> >>>> Artem >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/0891a714/attachment-0002.html From errotan at gmail.com Wed Dec 2 11:38:40 2009 From: errotan at gmail.com (=?utf-8?q?Pusk=C3=A1s_Zsolt?=) Date: Wed, 2 Dec 2009 20:38:40 +0100 Subject: [Freeswitch-users] CDR records In-Reply-To: <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> Message-ID: <200912022038.41102.errotan@gmail.com> 2009. december 2. 18.35.06 Michael Collins d?tummal ezt ?rta: > 2009/12/2 Jo?o Mesquita > > > What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid > > confusions. > > > > JM > > > > Thanks for catching my typo! :) > > -MC > Thanks for the hint I loaded mod_xml_cdr and now understand why there are 3 cdr records. I love mod_xml_cdr :) Thx! From anthony.minessale at gmail.com Wed Dec 2 11:46:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 13:46:31 -0600 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: Message-ID: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> you must only have 8k sounds so the resample is when it's playing files try make hd-sounds-install to install 16k sounds too On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: > Hi, I got a weird issue when I dialed an extension and listen to a recorded > voice mail greeting message. > After playing a couple of time of the greeting, the FS printed the warning > of "sample rate not matching", then > send the audio to a different remote RTP port. See the log below, > > > 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message.wav] (en:en) > 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate > doesn't match > 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec > Activated > 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from > xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 > 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original > codec. > 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less > than minimum record length: 3, discarding it. > 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-too-small.wav] (en:en) > 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ > > > the original codec is wideband 16kHz Speex and the wireshark shows that the > FS used the same codec. I used FS 1.04 in fedora 8. > I have two questions here, > (1) why does FS report "Sample rate doesn't match"? is it a bug or > configuration issue? > (2) Why does FS change the RTP port ? how to fix it? > > Thanks, > > Regards, > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/1b5e18de/attachment-0002.html From msc at freeswitch.org Wed Dec 2 12:06:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 12:06:27 -0800 Subject: [Freeswitch-users] CDR records In-Reply-To: <200912022038.41102.errotan@gmail.com> References: <200911291906.51520.errotan@gmail.com> <87f2f3b90912020935r747d0b0cg5f13bf44873d578e@mail.gmail.com> <200912022038.41102.errotan@gmail.com> Message-ID: <87f2f3b90912021206x1349a3belca432fa22ca961bf@mail.gmail.com> > I love mod_xml_cdr :) > > My sentiments as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/5bb524c4/attachment-0002.html From davis.erwin at gmail.com Wed Dec 2 12:08:37 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 15:08:37 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: Hi, Anthony, Thanks for your reply. When I type the command below, I got the error, Unknown target hd-sound-install make[1]: *** [hd-sound-install] Error 1 make: *** [hd-sound-install] Error 2 I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should explain why at first FS played in right sample rate. But after playing serveral time, FS complained about sample rate not matching. Any clue? Thanks, On 12/2/09, Anthony Minessale wrote: > > you must only have 8k sounds so the resample is when it's playing files > > try make hd-sounds-install to install 16k sounds too > > > > > > On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: > >> Hi, I got a weird issue when I dialed an extension and listen to a >> recorded voice mail greeting message. >> After playing a couple of time of the greeting, the FS printed the warning >> of "sample rate not matching", then >> send the audio to a different remote RTP port. See the log below, >> >> >> 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language >> specified - Using [en] >> 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle >> play-file:[voicemail/vm-record_message.wav] (en:en) >> 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing >> file >> 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >> [TRANSCODING_NECESSARY] >> 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate >> doesn't match >> 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec >> Activated >> 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port >> from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 >> 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore >> original codec. >> 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less >> than minimum record length: 3, discarding it. >> 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language >> specified - Using [en] >> 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle >> play-file:[voicemail/vm-too-small.wav] (en:en) >> 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec >> Activated L16 at 16000hz 1 channels 20ms >> 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 >> sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ >> >> >> the original codec is wideband 16kHz Speex and the wireshark shows that >> the FS used the same codec. I used FS 1.04 in fedora 8. >> I have two questions here, >> (1) why does FS report "Sample rate doesn't match"? is it a bug or >> configuration issue? >> (2) Why does FS change the RTP port ? how to fix it? >> >> Thanks, >> >> Regards, >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/9f8ee643/attachment-0002.html From kristian.kielhofner at gmail.com Wed Dec 2 12:16:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 2 Dec 2009 15:16:38 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> Message-ID: <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> Tony, Thanks for that but now it appears that the call just gets hung up on when the caller takes the callee off hold. Debug here: http://pastebin.freeswitch.org/11359 Thanks again! On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale wrote: > I decided to just change the code so its more elegant to handle recursive > broadcasting so you can try again and see if that helps. > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Dec 2 12:21:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 14:21:14 -0600 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: <191c3a030912021221r56ed768ei16dfe448d51a4a23@mail.gmail.com> that was make hd-sounds-install sorrry you should also update to SVN trunk because based on the line number in your log its clear you are using a much older version of FS On Wed, Dec 2, 2009 at 2:08 PM, Erwin Davis wrote: > Hi, Anthony, > > Thanks for your reply. > > When I type the command below, I got the error, > Unknown target hd-sound-install > make[1]: *** [hd-sound-install] Error 1 > make: *** [hd-sound-install] Error 2 > > I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, > there are directories, 8000, 16000, 32000, 48000 for recorded voicemail > greetings. It should explain why at first FS played in right sample rate. > But after playing serveral time, FS complained about sample rate not > matching. Any clue? Thanks, > > > > > > On 12/2/09, Anthony Minessale wrote: >> >> you must only have 8k sounds so the resample is when it's playing files >> >> try make hd-sounds-install to install 16k sounds too >> >> >> >> >> >> On Wed, Dec 2, 2009 at 12:41 PM, Erwin Davis wrote: >> >>> Hi, I got a weird issue when I dialed an extension and listen to a >>> recorded voice mail greeting message. >>> After playing a couple of time of the greeting, the FS printed the >>> warning of "sample rate not matching", then >>> send the audio to a different remote RTP port. See the log below, >>> >>> >>> 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing >>> file >>> 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language >>> specified - Using [en] >>> 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle >>> play-file:[voicemail/vm-record_message.wav] (en:en) >>> 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done >>> playing file >>> 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message >>> [TRANSCODING_NECESSARY] >>> 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate >>> doesn't match >>> 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec >>> Activated >>> 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port >>> from xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 >>> 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore >>> original codec. >>> 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less >>> than minimum record length: 3, discarding it. >>> 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language >>> specified - Using [en] >>> 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle >>> play-file:[voicemail/vm-too-small.wav] (en:en) >>> 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec >>> Activated L16 at 16000hz 1 channels 20ms >>> 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 >>> sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ >>> >>> >>> the original codec is wideband 16kHz Speex and the wireshark shows that >>> the FS used the same codec. I used FS 1.04 in fedora 8. >>> I have two questions here, >>> (1) why does FS report "Sample rate doesn't match"? is it a bug or >>> configuration issue? >>> (2) Why does FS change the RTP port ? how to fix it? >>> >>> Thanks, >>> >>> Regards, >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/31bfbd81/attachment-0002.html From gmartinr82 at gmail.com Wed Dec 2 10:04:58 2009 From: gmartinr82 at gmail.com (Martin Rodriguez) Date: Wed, 2 Dec 2009 15:04:58 -0300 Subject: [Freeswitch-users] First steps in FreeSWITCH Message-ID: Hi list; I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and 10 years in VoIP (Cisco). I need a reference guide to start working with FreeSWITCH. I download the official documentation, it would need some other configuration examples and dialplan sip device for calls. Martin Rodriguez From msc at freeswitch.org Wed Dec 2 12:57:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 12:57:18 -0800 Subject: [Freeswitch-users] First steps in FreeSWITCH In-Reply-To: References: Message-ID: <87f2f3b90912021257y6b2bea48lfb31a15ce151fcf@mail.gmail.com> On Wed, Dec 2, 2009 at 10:04 AM, Martin Rodriguez wrote: > Hi list; > > I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and > 10 years in VoIP (Cisco). I need a reference guide to start working > with > FreeSWITCH. I download the official documentation, it would need some > other configuration examples and dialplan sip device for calls. > > Martin Rodriguez > > Welcome to FreeSWITCH! We think you'll like it. Just remember that it's different than Cisco and Asterisk. (By "different" I mean "more powerful, better thought out, more flexible, and way, WAY cooler" :) ) First off you might want to peruse the Rosetta Stone: http://wiki.freeswitch.org/wiki/Rosetta_stone It's a place where you can get a frame of reference. If you've worked with Asterisk for six years then you've got lots of knowledge, and probably a few battle scars :) and that document helps you put things into perspective a bit. If you want a gentle intro to installing and setting up FS then check out this article: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT Note: the article mentions version 1.0.4 but we're really close to 1.0.5. We strongly recommend everyone, especially experienced users, to use the latest SVN trunk as it is almost always the most stable version of FS. Lastly, be sure to check out the getting started guide and the installation guide on the wiki. (www.freeswitch.org) The quick-and-dirty install guide is good for those who now Linux and just want to jump right in. -MC (IRC: mercutioviz) P.S. - we have a weekly FS community conf call and you're welcome to join: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/41ff6e43/attachment-0002.html From msc at freeswitch.org Wed Dec 2 13:03:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Dec 2009 13:03:02 -0800 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> Message-ID: <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: > Hi, Anthony, > > Thanks for your reply. > > When I type the command below, I got the error, > Unknown target hd-sound-install > make[1]: *** [hd-sound-install] Error 1 > make: *** [hd-sound-install] Error 2 > > I found out that under /usr/local/freeswitch/sounds/en/us/callie/voicemail, > there are directories, 8000, 16000, 32000, 48000 for recorded voicemail > greetings. It should explain why at first FS played in right sample rate. > But after playing serveral time, FS complained about sample rate not > matching. Any clue? Thanks, > > Erwin, As Tony said you've actually got a pretty old installation. If this is in production then I would recommend getting a sandbox machine, install trunk using the quick-and-dirty install, and then update the default config to you specific configuration. Test to make sure it works before you put it into production. :) Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run into any issues that require more real-time conversation. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/45cd76fa/attachment-0002.html From davis.erwin at gmail.com Wed Dec 2 13:15:17 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Wed, 2 Dec 2009 16:15:17 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> Message-ID: Hi, Anthony and Mike, Thanks for your reply. The problem still exists even after I ran "make hd-sounds install". I will try the latest version from the SVN to see if the problem will go away. I will let you know. Thanks folks, Regards, On 12/2/09, Michael Collins wrote: > > > > On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: > >> Hi, Anthony, >> >> Thanks for your reply. >> >> When I type the command below, I got the error, >> Unknown target hd-sound-install >> make[1]: *** [hd-sound-install] Error 1 >> make: *** [hd-sound-install] Error 2 >> >> I found out that under >> /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, >> 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should >> explain why at first FS played in right sample rate. But after playing >> serveral time, FS complained about sample rate not matching. Any clue? >> Thanks, >> >> > Erwin, > > As Tony said you've actually got a pretty old installation. If this is in > production then I would recommend getting a sandbox machine, install trunk > using the quick-and-dirty install, and then update the default config to you > specific configuration. Test to make sure it works before you put it into > production. :) > > Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run > into any issues that require more real-time conversation. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/05b03e36/attachment-0002.html From larclap at yahoo.com Wed Dec 2 14:07:38 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 14:07:38 -0800 Subject: [Freeswitch-users] Eavesdrop error? Message-ID: <036401ca739b$dc451340$94cf39c0$@com> I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars From abeka at greatiam.com Wed Dec 2 14:44:42 2009 From: abeka at greatiam.com (Otis) Date: Wed, 02 Dec 2009 22:44:42 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <20091202175853.GW31924@base.carmickle.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> Message-ID: <4B16EDDA.5040908@greatiam.com> Thanks. Will let you know Frank Carmickle wrote: > On Wed, Dec 02, Otis wrote: > Snip... > > >> Thanks. >> >> I would like all extensions on say server A to be contactable by those >> on server B and vice versa. >> > > The example I gave you should get you started. Let us know how you get along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > > From larclap at yahoo.com Wed Dec 2 15:22:20 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 15:22:20 -0800 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <036401ca739b$dc451340$94cf39c0$@com> References: <036401ca739b$dc451340$94cf39c0$@com> Message-ID: <03b101ca73a6$4c8db080$e5a91180$@com> Sorry, svn 15753 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Wednesday, December 02, 2009 2:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Eavesdrop error? I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Wed Dec 2 15:33:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 17:33:22 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> Message-ID: <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> I am not sure what you are sending over the socket but you have a queued hangup being processed on line 640 of your pastebin are you executing any commands with a ! character in it by any chance or executing the hangup app on purpose? On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Tony, > > Thanks for that but now it appears that the call just gets hung up > on when the caller takes the callee off hold. Debug here: > > http://pastebin.freeswitch.org/11359 > > Thanks again! > > On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale > wrote: > > I decided to just change the code so its more elegant to handle recursive > > broadcasting so you can try again and see if that helps. > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/1dad3d49/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 2 15:34:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Dec 2009 17:34:57 -0600 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03b101ca73a6$4c8db080$e5a91180$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> Message-ID: <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> it probably just means the uuid was not retrieved from the db when you called the eavesdrop exten which does the lookup on the uuid for the hash key based on what ext you hit to retrieve the most recent uuid that called that ext. On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > Sorry, svn 15753 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars > Zeb > Sent: Wednesday, December 02, 2009 2:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Eavesdrop error? > > I tried to use eavesdrop today and it did not work. The error message in > the > log is: > > [ERR] mod_dptools.c:334 Usage: [all | ] > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > incorrect? > > http://pastebin.freeswitch.org/11363 > > Thanks Lars > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/95347fd4/attachment-0002.html From erandr-junk at usa.net Wed Dec 2 16:31:31 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 16:31:31 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> Message-ID: <26619085.post@talk.nabble.com> Can I reduce resolution of that timer thread 10 times? I mean, I glanced through the code, and see that among others (are there others?) RTP and IVR set up their timers that are subsequently managed by this thread. RTP timers should be eliminated by that setting you've suggested. IVR timers are set at 20ms... So, if the thread is set to wake up every 10ms instead of 1ms it should be able to wake up those IVR timers just fine. Right? That's a cool design to have one dedicated thread that maintains accurate timing and then broadcasts via condition variables to hundreds of other threads events that they can register for. I'm sure it's one of the reasons why FS scales so much better than Asterisk. But for poor low-end setups that sit in the closet, eat only 6W of power and hardly ever run more than two calls at the same time, can I hack it somehow to be more UNIX-friendly? I.e. make it stuck in select() or recv() when there is nothing to do, call clock_gettime() right from the thread that wants and when it wants to know current time? Say, what if that thread is made to suspend on a condition variable in case if there are no timers registered in TIMER_MATRIX? Then, if some other thread comes up and adds its timer into the matrix, it could wake up the timer thread and enjoy accurate timing as needed, on demand? And in-between the calls, when there is no RTP or IVR, it will all go silent? I mean, sitting on a wait queue in the kernel is way better than go back and forth incrementing counters that nobody even needs at the moment? Anthony Minessale-2 wrote: > > idle is a 4 letter word to a realtime application. > > The core keeps a single high-priority thread to keep 1ms timing and > expands > that broadcasting > to hundreds or thousand of threads who need accurate timing. > > Your choppy audio is caused by linksys lying about the packet len that > it's > using and we set our timer > to the wrong speed. > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From timuckun at gmail.com Wed Dec 2 16:49:36 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 3 Dec 2009 13:49:36 +1300 Subject: [Freeswitch-users] HA questions. Message-ID: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to "lose" those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. From larclap at yahoo.com Wed Dec 2 18:19:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Dec 2009 18:19:58 -0800 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> Message-ID: <03c401ca73bf$1cea8600$56bf9200$@com> Is this reasonable given it was the only call in FreeSwitch at the time? How can this situation be corrected in the future? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, December 02, 2009 3:35 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Eavesdrop error? it probably just means the uuid was not retrieved from the db when you called the eavesdrop exten which does the lookup on the uuid for the hash key based on what ext you hit to retrieve the most recent uuid that called that ext. On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: Sorry, svn 15753 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Wednesday, December 02, 2009 2:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Eavesdrop error? I tried to use eavesdrop today and it did not work. The error message in the log is: [ERR] mod_dptools.c:334 Usage: [all | ] I simply dialed 881010, trying to eavesdrop on extension 1010. Is this incorrect? http://pastebin.freeswitch.org/11363 Thanks Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From erandr-junk at usa.net Wed Dec 2 19:35:39 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 19:35:39 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26619085.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> Message-ID: <26620518.post@talk.nabble.com> Oh, looks like the timers are also used for streaming local data in read_stream_thread(). Due to this there is always one timer active with 20ms interval. But wait a sec, why is freeswitch periodically trying to stream /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav somewhere? Every minute or so? Did I misconfigure it? eaf wrote: > > Say, what if that thread is made to suspend on a condition variable in > case if there are no timers registered in TIMER_MATRIX? Then, if some > other thread comes up and adds its timer into the matrix, it could wake up > the timer thread and enjoy accurate timing as needed, on demand? And > in-between the calls, when there is no RTP or IVR, it will all go silent? > I mean, sitting on a wait queue in the kernel is way better than go back > and forth incrementing counters that nobody even needs at the moment? > > > Anthony Minessale-2 wrote: >> >> idle is a 4 letter word to a realtime application. >> >> The core keeps a single high-priority thread to keep 1ms timing and >> expands >> that broadcasting >> to hundreds or thousand of threads who need accurate timing. >> >> Your choppy audio is caused by linksys lying about the packet len that >> it's >> using and we set our timer >> to the wrong speed. >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620518.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at psrm.ca Wed Dec 2 19:35:59 2009 From: john at psrm.ca (John Lalande) Date: Wed, 2 Dec 2009 21:35:59 -0600 Subject: [Freeswitch-users] can't register Inphonex Message-ID: <005a01ca73c9$bc2dcf60$34896e20$@ca> I am new to FS having ditched Asterisk a few weeks ago. I have iptel registered but I cannot get Inphonex to work. I am using the settings from http://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no avail. The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout [408]." Is there some way to debug this? sofia status displays: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at 192.168.125.15:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG inphonex gateway sip:5285418 at sip.inphonex.com FAILED (retry: 28s) iptel gateway sip:jlalande at sip.iptel.org REGED internal profile sip:mod_sofia at 192.168.125.15:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) 192.168.125.15 alias internal ALIASED ============================================================================ ===================== -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091202/43b7aa50/attachment-0002.html From erandr-junk at usa.net Wed Dec 2 19:47:42 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 19:47:42 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26620518.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <26620518.post@talk.nabble.com> Message-ID: <26620610.post@talk.nabble.com> OK, I'm slow. It's music-on-hold, and it's playing non-stop like that timer thread. Even when there are no calls. Why? eaf wrote: > > Oh, looks like the timers are also used for streaming local data in > read_stream_thread(). Due to this there is always one timer active with > 20ms interval. > > But wait a sec, why is freeswitch periodically trying to stream > /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav somewhere? > Every minute or so? Did I misconfigure it? > > > eaf wrote: >> >> Say, what if that thread is made to suspend on a condition variable in >> case if there are no timers registered in TIMER_MATRIX? Then, if some >> other thread comes up and adds its timer into the matrix, it could wake >> up the timer thread and enjoy accurate timing as needed, on demand? And >> in-between the calls, when there is no RTP or IVR, it will all go silent? >> I mean, sitting on a wait queue in the kernel is way better than go back >> and forth incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620610.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Dec 2 19:57:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Dec 2009 22:57:06 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26620518.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <26620518.post@talk.nabble.com> Message-ID: <3377EB5F-5022-4D72-B56C-61B3ED25A304@jerris.com> This is keeping track of a place in the music on hold so your hold music does not start back up at the same place every time. If you don't want to do this it is a module that you don't need to load and you can get your moh from any soundfile at your choice in configuration. Mike On Dec 2, 2009, at 10:35 PM, eaf wrote: > > Oh, looks like the timers are also used for streaming local data in > read_stream_thread(). Due to this there is always one timer active > with 20ms > interval. > > But wait a sec, why is freeswitch periodically trying to stream > /opt/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav > somewhere? > Every minute or so? Did I misconfigure it? > > > eaf wrote: >> >> Say, what if that thread is made to suspend on a condition variable >> in >> case if there are no timers registered in TIMER_MATRIX? Then, if some >> other thread comes up and adds its timer into the matrix, it could >> wake up >> the timer thread and enjoy accurate timing as needed, on demand? And >> in-between the calls, when there is no RTP or IVR, it will all go >> silent? >> I mean, sitting on a wait queue in the kernel is way better than go >> back >> and forth incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len >>> that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26620518.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Wed Dec 2 20:00:00 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 2 Dec 2009 23:00:00 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26619085.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> Message-ID: <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> In short. No, you can not for many reasons. The milisecond tic is used throughout the code even when there is not any calls up. You can grep for switch_cond_next if you would like to see where but it is required to keep our global timestamp and for pacing the scheduler among other services that run all the time. Mike On Dec 2, 2009, at 7:31 PM, eaf wrote: > > Can I reduce resolution of that timer thread 10 times? I mean, I > glanced > through the code, and see that among others (are there others?) RTP > and IVR > set up their timers that are subsequently managed by this thread. > RTP timers > should be eliminated by that setting you've suggested. IVR timers > are set at > 20ms... So, if the thread is set to wake up every 10ms instead of > 1ms it > should be able to wake up those IVR timers just fine. Right? > > That's a cool design to have one dedicated thread that maintains > accurate > timing and then broadcasts via condition variables to hundreds of > other > threads events that they can register for. I'm sure it's one of the > reasons > why FS scales so much better than Asterisk. But for poor low-end > setups that > sit in the closet, eat only 6W of power and hardly ever run more > than two > calls at the same time, can I hack it somehow to be more UNIX- > friendly? I.e. > make it stuck in select() or recv() when there is nothing to do, call > clock_gettime() right from the thread that wants and when it wants > to know > current time? > > Say, what if that thread is made to suspend on a condition variable > in case > if there are no timers registered in TIMER_MATRIX? Then, if some other > thread comes up and adds its timer into the matrix, it could wake up > the > timer thread and enjoy accurate timing as needed, on demand? And in- > between > the calls, when there is no RTP or IVR, it will all go silent? I mean, > sitting on a wait queue in the kernel is way better than go back and > forth > incrementing counters that nobody even needs at the moment? > > > Anthony Minessale-2 wrote: >> >> idle is a 4 letter word to a realtime application. >> >> The core keeps a single high-priority thread to keep 1ms timing and >> expands >> that broadcasting >> to hundreds or thousand of threads who need accurate timing. >> >> Your choppy audio is caused by linksys lying about the packet len >> that >> it's >> using and we set our timer >> to the wrong speed. >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From erandr-junk at usa.net Wed Dec 2 20:58:39 2009 From: erandr-junk at usa.net (eaf) Date: Wed, 2 Dec 2009 20:58:39 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> Message-ID: <26621005.post@talk.nabble.com> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides that. Yeah, there is a global timestamp... It's easy to workaround that for RTP who calls switch_micro_time_now()... But if somebody accesses runtime.timestamp directly, it's gonna be tough to grep for that. If only this was C++... I'll play around. Never liked polling too much. Never could've guessed that polling could be so useful for scalability ;) My naive implementation would've pulled timestamp via system calls and would've done sleeping by passing exact interval to select() instead of syncing with a pacing thread. Which would be dead-quiet at idle time, but, of course, would stop scaling at some point due to excessive number of system calls. Thanks. Michael Jerris wrote: > > In short. No, you can not for many reasons. The milisecond tic is > used throughout the code even when there is not any calls up. You can > grep for switch_cond_next if you would like to see where but it is > required to keep our global timestamp and for pacing the scheduler > among other services that run all the time. > > Mike > > On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >> Can I reduce resolution of that timer thread 10 times? I mean, I >> glanced >> through the code, and see that among others (are there others?) RTP >> and IVR >> set up their timers that are subsequently managed by this thread. >> RTP timers >> should be eliminated by that setting you've suggested. IVR timers >> are set at >> 20ms... So, if the thread is set to wake up every 10ms instead of >> 1ms it >> should be able to wake up those IVR timers just fine. Right? >> >> That's a cool design to have one dedicated thread that maintains >> accurate >> timing and then broadcasts via condition variables to hundreds of >> other >> threads events that they can register for. I'm sure it's one of the >> reasons >> why FS scales so much better than Asterisk. But for poor low-end >> setups that >> sit in the closet, eat only 6W of power and hardly ever run more >> than two >> calls at the same time, can I hack it somehow to be more UNIX- >> friendly? I.e. >> make it stuck in select() or recv() when there is nothing to do, call >> clock_gettime() right from the thread that wants and when it wants >> to know >> current time? >> >> Say, what if that thread is made to suspend on a condition variable >> in case >> if there are no timers registered in TIMER_MATRIX? Then, if some other >> thread comes up and adds its timer into the matrix, it could wake up >> the >> timer thread and enjoy accurate timing as needed, on demand? And in- >> between >> the calls, when there is no RTP or IVR, it will all go silent? I mean, >> sitting on a wait queue in the kernel is way better than go back and >> forth >> incrementing counters that nobody even needs at the moment? >> >> >> Anthony Minessale-2 wrote: >>> >>> idle is a 4 letter word to a realtime application. >>> >>> The core keeps a single high-priority thread to keep 1ms timing and >>> expands >>> that broadcasting >>> to hundreds or thousand of threads who need accurate timing. >>> >>> Your choppy audio is caused by linksys lying about the packet len >>> that >>> it's >>> using and we set our timer >>> to the wrong speed. >>> >>> >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26621005.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Wed Dec 2 21:11:59 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 07:11:59 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO Message-ID: Hello, I have Polycom phones which send only RFC-2833 (or inband which I dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco gateway has some bug and accepts only INFO. I did a few tests: - Some of the phones are on different profile than the Cisco. On their profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set 'dtmf-type=info' and Freeswitch did the translation. All works ok... - Some of the phones are on the same profile as the Cisco, so I must set dtmf-type to rfc2833; it works with internal applications (like voicemail) but does not work through the Cisco as it misinterprets the rfc2833 Is there a way to set some variable (or a parameter to the bridge application) to do the translation? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/7160286f/attachment-0002.html From jingwei.yang at gmail.com Wed Dec 2 22:02:47 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:02:47 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c Message-ID: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function ???X11_errors_handler???: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ???skypiax_send_message???: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/d0dab081/attachment-0002.html From mrene_lists at avgs.ca Wed Dec 2 22:09:53 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Dec 2009 01:09:53 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> Message-ID: <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > Hi Guys, > > I got a compilation error of skypiax_protocol.c with the latest > version r15764. > > Compiling skypiax_protocol.c... > cc1: warnings being treated as errors > skypiax_protocol.c: In function ???X11_errors_handler???: > skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c: In function ???skypiax_send_message???: > skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: > skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations > and code > skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations > and code > make[5]: *** [skypiax_protocol.o] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_skypiax-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > I personally checked the file and it shouldn't be a merge problem. > Does anyone encounter this as well? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8a36213f/attachment-0002.html From jingwei.yang at gmail.com Wed Dec 2 22:33:16 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:33:16 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> Message-ID: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > Consider it fixed. > Committed revision 15765. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > > Hi Guys, > > I got a compilation error of skypiax_protocol.c with the latest version > r15764. > > Compiling skypiax_protocol.c... > *cc1: warnings being treated as errors* > skypiax_protocol.c: In function ???X11_errors_handler???: > skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c: In function ???skypiax_send_message???: > skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: > skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and > code > skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and > code > make[5]: *** [skypiax_protocol.o] Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_skypiax-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > I personally checked the file and it shouldn't be a merge problem. Does > anyone encounter this as well? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b632a9a9/attachment-0002.html From mrene_lists at avgs.ca Wed Dec 2 22:39:16 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Dec 2009 01:39:16 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Message-ID: <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG - > std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings - > Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden - > DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o > make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/ > tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/ > libtiff.a -ljpeg -lz -lm -lc > ./make_at_dictionary: error while loading shared libraries: > libjpeg.so.7: cannot open shared object file: No such file or > directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene > wrote: > Consider it fixed. > Committed revision 15765. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: > >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest >> version r15764. >> >> Compiling skypiax_protocol.c... >> cc1: warnings being treated as errors >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func? >> ??: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed >> declarations and code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed >> declarations and code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. >> Does anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c91d9e51/attachment-0002.html From jingwei.yang at gmail.com Wed Dec 2 22:43:30 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:43:30 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> Message-ID: <13529f9d0912022243y700728d4l30c7eb4e3152d1c9@mail.gmail.com> Not sure whether this error is due to the lack of libjpeg. I just double checked, this library had been installed. Package libjpeg-6b-37.i386 already installed and latest version On Thu, Dec 3, 2009 at 2:33 PM, Jingwei Yang wrote: > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc > ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: > cannot open shared object file: No such file or directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest version >> r15764. >> >> Compiling skypiax_protocol.c... >> *cc1: warnings being treated as errors* >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >> code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. Does >> anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/66b655b8/attachment-0002.html From yehavi.bourvine at gmail.com Wed Dec 2 22:48:12 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 08:48:12 +0200 Subject: [Freeswitch-users] How do I know the destination profile name? In-Reply-To: References: <4B0266F4.8070602@savion.huji.ac.il> <191c3a030911231157p44612c5dm3f0ee1e7b37e9cd3@mail.gmail.com> Message-ID: BTW, I forgot to update: I changed the bridge parameters to use sofia_contact() and it solved the problem. I also fixed the presence problem I had before with sofia_contact() (added presence_id to the bridge command). Regards, __Yehavi: 2009/11/24 Yehavi Bourvine > Hello Anthony, > > Indeed I see the reference to this channel variable in the code, but when > trying to access it from the dial plan it is empty... I try to get the value > of ${sip_profile_name} and it is empty. > > Thanks! __Yehavi: > > 2009/11/23 Anthony Minessale > >> Let's just do this: >> >> r15629 or higher >> >> look for sip_profile_name >> >> >> >> >> On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun wrote: >> >>> Hi >>> We have more then one profile. To make a call I have to enter : bridge >>> sofia/profile/number at ip >>> The problem is when I use : "${use_profile}" I am getting the caller >>> profile, and I need the destination profile. >>> >>> How do I get this information? >>> >>> Thanks >>> >>> Eli >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b271714d/attachment-0002.html From jingwei.yang at gmail.com Wed Dec 2 22:49:35 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 14:49:35 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> Message-ID: <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > Hi, > > That one is on your side. If you changed/updated system libs it might be > worth doing another ./configure > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > > Hi Mathieu, thanks for the promptly reply. The error has been fixed. > However, I encounter another one. > > gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc > ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: > cannot open shared object file: No such file or directory > make[8]: *** [at_interpreter_dictionary.h] Error 127 > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > Do you have idea about this one? > > Thanks! > > On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: > >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >> Hi Guys, >> >> I got a compilation error of skypiax_protocol.c with the latest version >> r15764. >> >> Compiling skypiax_protocol.c... >> *cc1: warnings being treated as errors* >> skypiax_protocol.c: In function ???X11_errors_handler???: >> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_send_message???: >> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >> code >> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >> code >> make[5]: *** [skypiax_protocol.o] Error 1 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_skypiax-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> I personally checked the file and it shouldn't be a merge problem. Does >> anyone encounter this as well? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/74598504/attachment-0002.html From devel at thom.fr.eu.org Thu Dec 3 00:00:55 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 09:00:55 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail In-Reply-To: <20091202181527.GY31924@base.carmickle.com> References: <20091202181527.GY31924@base.carmickle.com> Message-ID: <9b3453c2ea2580e6080cfb4dbd6e3d7e@thom.fr.eu.org> Well, I'm just starting to use freeswitch, so my approach is probably for from optimal. The point is I wanted that voicemail do not prompt for passwords when the caller is a sip registered user, but I also wanted the login requirement if the voicemail was called from some FXS port. That lead me to having : in my dialplan. Fran?ois On Wed, 2 Dec 2009 13:15:28 -0500, Frank Carmickle wrote: > On Wed, Dec 02, Fran??ois Legal wrote: >> No, my voicemail extension (I have 2 actually) is called >> vmain_unregistered_user, so in voicemail.conf.xml I have : > > Also, is there a functional requirement for two different extensions to > call vmain? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jingwei.yang at gmail.com Thu Dec 3 00:24:04 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 16:24:04 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> Message-ID: <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ???command_search???: at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in this function) at_interpreter.c: In function ???at_interpreter???: at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > No, I didn't change or update the system libs. I just wanted to double > check whether my system has this libjpeg library. ./configure was definitely > executed before the source codes were rebuilt. > > Regards, > -Jingwei > > > On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > >> Hi, >> >> That one is on your side. If you changed/updated system libs it might be >> worth doing another ./configure >> >> Cheers, >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >> >> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >> However, I encounter another one. >> >> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> -lc >> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: >> cannot open shared object file: No such file or directory >> make[8]: *** [at_interpreter_dictionary.h] Error 127 >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> Do you have idea about this one? >> >> Thanks! >> >> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >> >>> Consider it fixed. >>> Committed revision 15765. >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>> >>> Hi Guys, >>> >>> I got a compilation error of skypiax_protocol.c with the latest version >>> r15764. >>> >>> Compiling skypiax_protocol.c... >>> *cc1: warnings being treated as errors* >>> skypiax_protocol.c: In function ???X11_errors_handler???: >>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ???skypiax_send_message???: >>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >>> code >>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >>> code >>> make[5]: *** [skypiax_protocol.o] Error 1 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_skypiax-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> I personally checked the file and it shouldn't be a merge problem. Does >>> anyone encounter this as well? >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b6d1f51c/attachment-0002.html From oseslija at gmail.com Thu Dec 3 00:43:28 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 3 Dec 2009 09:43:28 +0100 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: Message-ID: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Bear in mind that FS will accept both 2833 and INFO in any profile on an inbound call. Param "dtmf-type" is valid only for outbound calls from the profile. Ognjen On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine wrote: > Hello, > > I have Polycom phones which send only RFC-2833 (or inband which I > dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco > gateway has some bug and accepts only INFO. > > I did a few tests: > > - Some of the phones are on different profile than the Cisco. On their > profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set > 'dtmf-type=info' and Freeswitch did the translation. All works ok... > - Some of the phones are on the same profile as the Cisco, so I must > set dtmf-type to rfc2833; it works with internal applications (like > voicemail) but does not work through the Cisco as it misinterprets the > rfc2833 > > > Is there a way to set some variable (or a parameter to the bridge > application) to do the translation? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/dbcf0a36/attachment-0002.html From kond at nstel.ru Thu Dec 3 00:44:31 2009 From: kond at nstel.ru (Nikolay Kondratyev) Date: Thu, 3 Dec 2009 11:44:31 +0300 Subject: [Freeswitch-users] call barge in In-Reply-To: <87f2f3b90912021002p3db48213l9dc666774fa9ab25@mail.gmail.com> Message-ID: <20091203084431.DC45D11A9D@mail.nstel.ru> Michael, Mark, Artem, Thank you for your answers. I believe FS will suite our needs. I've installed dedicated virtual machine (Centos) for FS and going to play with it. Thanks and regards, Nikolay. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 02, 2009 9:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] call barge in On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov wrote: 1 - config 2 - I've done this with programming 3 - suppose programming would be needed Just to clarify, when you say "programming" there are different levels of involvement. For example, you can do programming in C which is pretty in depth, but that's probably not what is required. Most likely this all can be done with dialplan configuration and some simple Lua/Perl/JavaScript scripts. (We support many scripting languages.) I recommend that you install FreeSWITCH on a test server and connect a few phones. Start with the default configuration and make sure that you have it working properly and go from there. Also, we have an IRC channel on irc.freenode.net where you can come and discuss things realtime. Lastly, we have a weekly conference call where you can ask community members and developers your questions: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call I recommend that you use the latest SVN trunk as we are really close to 1.0.5. If you're on a Linux box you can do the quick install process mentioned here: http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install Dive in and have fun! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/af25ea7a/attachment-0002.html From jingwei.yang at gmail.com Thu Dec 3 01:29:08 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Thu, 3 Dec 2009 17:29:08 +0800 Subject: [Freeswitch-users] OpenZap issues with incoming and outgoing calls Message-ID: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> Hello All, I have a Digium TDM400P pci card with two FXO ports installed on my linux box. I've connected an external telephone line to the first FXO port. But I can't either make outgoing calls or receive incoming ones. Here are my setups, please let me know where goes wrong. * /etc/zaptel.conf* loadzone = sg defaultzone=sg fxsks=1,2 */usr/local/freeswitch/conf/zt.conf* remains unchanged [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 */usr/local/freeswitch/conf/openzap.conf* [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* I defined an extension in dialplan/default.xml to receive bridge incoming calls to my skype instance. Frankly speaking, I'm not sure whether this definition is correct. How should I define the expression? When I dial the telephone number, the FS console has no response and I hear nother but busy tones. For outgoing calls, I tried something like this: originate openzap/1/1/xxxxxxxx &echo, while "xxxxxxxx" is my handphone number. Again, my handphone has no response. Hopefully I've explained my situation clearly. Please kindly enlighten where the problem might be. Thanks, -Jingwei p.s. here is the outgoing log trace for your reference. freeswitch at localhost.localdomain> originate openzap/1/1/xxxxxxxx &echo 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound channel OpenZAP/1:1/xxxxxxxx 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel OpenZAP/1:1/xxxxxxxx [6f843194-18ce-4525-862f-f5f4e96db5eb] 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 (OpenZAP/1:1/xxxxxxxx) State Change CS_NEW -> CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 from DOWN to DIALING 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/xxxxxxxx) State Change CS_INIT -> CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/xxxxxxxx) State INIT going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/xxxxxxxx CHANNEL ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/1:1/xxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/xxxxxxxx) State ROUTING going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA going to sleep 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_HOME ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_OFFICE ||| 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: |||USER amanda8884 PHONE_MOBILE ||| 2009-12-03 17:21:34.684709 [DEBUG] ozmod_analog.c:340 Changing state on 1:1 from DIALING to BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for BUSY 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:579 Changing state on 1:1 from BUSY to DOWN 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DOWN 2009-12-03 17:21:34.724706 [DEBUG] mod_openzap.c:1334 got FXO sig 1:1 [STOP] 2009-12-03 17:21:34.724706 [NOTICE] mod_openzap.c:1352 Hangup OpenZAP/1:1/xxxxxxxx [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.724706 [DEBUG] switch_channel.c:1912 Send signal OpenZAP/1:1/xxxxxxxx [KILL] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_HANGUP 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:459 thread mismatch skipping state handler. 2009-12-03 17:21:34.724706 [DEBUG] zap_io.c:1234 channel done 1:1 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:766 ANALOG CHANNEL 1:1 thread ended. 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP API CALL [originate(openzap/1/1/xxxxxxxx &echo)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.724706 [DEBUG] switch_ivr_originate.c:2988 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] 2009-12-03 17:21:34.725762 [WARNING] mod_openzap.c:474 VETO Changing state on 1:1 from DOWN to HANGUP 2009-12-03 17:21:34.725762 [DEBUG] mod_openzap.c:510 OpenZAP/1:1/xxxxxxxx CHANNEL HANGUP 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:46 OpenZAP/1:1/xxxxxxxx Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:486 (OpenZAP/1:1/xxxxxxxx) State HANGUP going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/1:1/xxxxxxxx) State Change CS_HANGUP -> CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:53 OpenZAP/1:1/xxxxxxxx Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 (OpenZAP/1:1/xxxxxxxx) State REPORTING going to sleep 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/1:1/xxxxxxxx) State Change CS_REPORTING -> CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/xxxxxxxx [BREAK] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:1136 Session 1 (OpenZAP/1:1/xxxxxxxx) Locked, Waiting on external entities 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1154 Session 1 (OpenZAP/1:1/xxxxxxxx) Ended 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1156 Close Channel OpenZAP/1:1/xxxxxxxx [CS_DESTROY] 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:423 (OpenZAP/1:1/xxxxxxxx) Running State Change CS_DESTROY 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY freeswitch at localhost.localdomain> 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:60 OpenZAP/1:1/xxxxxxxx Standard DESTROY 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:434 (OpenZAP/1:1/xxxxxxxx) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/a22e8f5a/attachment-0002.html From b_ball_henry at hotmail.com Thu Dec 3 01:36:46 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 3 Dec 2009 17:36:46 +0800 Subject: [Freeswitch-users] Gateway issue with no audio Message-ID: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. Finally, I gave up play around with the internal profile and put the gateway *settings under external profile. And magically, it worked.* I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? by the way, I am using freeswitch 1.4 stable version. -- Henry Huang -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8d14e1f8/attachment-0002.html From devel at thom.fr.eu.org Thu Dec 3 02:16:25 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 11:16:25 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Message-ID: <82fda3419daad21bded177dc2b113396@thom.fr.eu.org> I'm already using the latest wanpipe drivers (3.5.8), so yes. Fran?ois On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitch.org [9] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [10] ClueCon http://www.cluecon.com/ [11] Twitter: http://twitter.com/FreeSWITCH_wire [12] AIM: anthm MSN:anthony_minessale at hotmail.com [13] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [14] IRC: irc.freenode.net [15] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [16] iax:guest at conference.freeswitch.org/888 [17] googletalk:conf+888 at conference.freeswitch.org [18] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [23] ClueCon http://www.cluecon.com/ [24] Twitter: http://twitter.com/FreeSWITCH_wire [25] AIM: anthm MSN:anthony_minessale at hotmail.com [26] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [27] IRC: irc.freenode.net [28] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [29] iax:guest at conference.freeswitch.org/888 [30] googletalk:conf+888 at conference.freeswitch.org [31] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [32] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [33] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [34] http://www.freeswitch.org [35] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [36] ClueCon http://www.cluecon.com/ [37] Twitter: http://twitter.com/FreeSWITCH_wire [38] AIM: anthm MSN:anthony_minessale at hotmail.com [39] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [40] IRC: irc.freenode.net [41] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [42] iax:guest at conference.freeswitch.org/888 [43] googletalk:conf+888 at conference.freeswitch.org [44] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:anthony.minessale at gmail.com [3] mailto:devel at thom.fr.eu.org [4] http://files.freeswitch.org [5] mailto:devel at thom.fr.eu.org [6] mailto:FreeSWITCH-users at lists.freeswitch.org [7] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] http://lists.freeswitch.org/mailman/options/freeswitch-users [9] http://www.freeswitch.org [10] http://www.freeswitch.org/ [11] http://www.cluecon.com/ [12] http://twitter.com/FreeSWITCH_wire [13] mailto:MSN%3Aanthony_minessale at hotmail.com [14] mailto:PAYPAL%3Aanthony.minessale at gmail.com [15] http://irc.freenode.net [16] mailto:sip%3A888 at conference.freeswitch.org [17] http://iax:guest at conference.freeswitch.org/888 [18] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [19] mailto:FreeSWITCH-users at lists.freeswitch.org [20] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21] http://lists.freeswitch.org/mailman/options/freeswitch-users [22] http://www.freeswitch.org [23] http://www.freeswitch.org/ [24] http://www.cluecon.com/ [25] http://twitter.com/FreeSWITCH_wire [26] mailto:MSN%3Aanthony_minessale at hotmail.com [27] mailto:PAYPAL%3Aanthony.minessale at gmail.com [28] http://irc.freenode.net [29] mailto:sip%3A888 at conference.freeswitch.org [30] http://iax:guest at conference.freeswitch.org/888 [31] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [32] mailto:FreeSWITCH-users at lists.freeswitch.org [33] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] http://lists.freeswitch.org/mailman/options/freeswitch-users [35] http://www.freeswitch.org [36] http://www.freeswitch.org/ [37] http://www.cluecon.com/ [38] http://twitter.com/FreeSWITCH_wire [39] mailto:MSN%3Aanthony_minessale at hotmail.com [40] mailto:PAYPAL%3Aanthony.minessale at gmail.com [41] http://irc.freenode.net [42] mailto:sip%3A888 at conference.freeswitch.org [43] http://iax:guest at conference.freeswitch.org/888 [44] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/a9643b6b/attachment-0002.html From devel at thom.fr.eu.org Thu Dec 3 02:17:09 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Thu, 03 Dec 2009 11:17:09 +0100 Subject: [Freeswitch-users] Remote fetching of voicemail Message-ID: <0d3ea0a7a75877ecc715076b03899b06@thom.fr.eu.org> Thanks. I did not succed to fincing the correct syntx with inline, but the transfer application did work. Fran?ois On Wed, 2 Dec 2009 12:21:54 -0600, Anthony Minessale wrote: bind to the transfer app so that it transfers the call to the vm extension that way the current application is always interrupted and replaced. The special "inline" dialplan lets you transfer calls right to an application use "inline" as the dp name and voicemail: as the extension On Wed, Dec 2, 2009 at 4:57 AM, Fran?ois Legal wrote: Hello, I created an extension in my dialplan so that when an incoming call arrives, it rings a group of lines and then fallback to the voicemail if no line is answered. I wanted then that when voicemail starts, the calling party could dial some numbers to fetch the voicemail. I used bind_meta_app for this. My problem is, when using bind_meta_app, the voicemail continues, and I sometimes experience freeswitch hanging after the call is over, depending on when the bind_meta_app is activated. How can I make freeswitch terminate the first voicemail instance when activating the bind_meta_app. Here's my extension : Thanks Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [4] http://www.freeswitch.org [5] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [6] ClueCon http://www.cluecon.com/ [7] Twitter: http://twitter.com/FreeSWITCH_wire [8] AIM: anthm MSN:anthony_minessale at hotmail.com [9] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [10] IRC: irc.freenode.net [11] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [12] iax:guest at conference.freeswitch.org/888 [13] googletalk:conf+888 at conference.freeswitch.org [14] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:FreeSWITCH-users at lists.freeswitch.org [3] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [4] http://lists.freeswitch.org/mailman/options/freeswitch-users [5] http://www.freeswitch.org [6] http://www.freeswitch.org/ [7] http://www.cluecon.com/ [8] http://twitter.com/FreeSWITCH_wire [9] mailto:MSN%3Aanthony_minessale at hotmail.com [10] mailto:PAYPAL%3Aanthony.minessale at gmail.com [11] http://irc.freenode.net [12] mailto:sip%3A888 at conference.freeswitch.org [13] http://iax:guest at conference.freeswitch.org/888 [14] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c138525c/attachment-0002.html From codecomplete at free.fr Thu Dec 3 04:17:22 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 3 Dec 2009 04:17:22 -0800 (PST) Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? Message-ID: <26625105.post@talk.nabble.com> Hello In a thread back in March, I read that support for IAX in FreeSwitch is a bit of kludge and since there's not much demand for it, chances are it won't improve in the foreseeable future. So I'd like some feedback from users who routinely connect to a FreeSwitch server from various venues, ie. wifi hotspots at McD, Ethernet LAN in hotels, etc. (in my case, the FreeSwitch server is located in a private network behind a NAT router with SIP/RTP ports statically mapped.) Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) ports fail being opened dynamically to work properly, or does SIP today really work well over NAT firewalls? Thank you. -- View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From oscav at hotmail.fr Thu Dec 3 04:21:04 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 3 Dec 2009 04:21:04 -0800 (PST) Subject: [Freeswitch-users] How to run a JS script periodically Message-ID: <26625147.post@talk.nabble.com> Hi, Someone knows how to run periodically a JS script ?? The purpose is to write to a db some global informations (Global Variables) about FS like every 5 minutes. Thanks. -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Thu Dec 3 05:28:57 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 3 Dec 2009 21:28:57 +0800 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <26625147.post@talk.nabble.com> References: <26625147.post@talk.nabble.com> Message-ID: <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> Not sure about js, but in lua, you can use luarun to run a long-running script like loop do sth. sleep 5min end and also it can be set to start with freeswitch in lua.conf.xml I guess you can also use jsrun to run js. And, if you run every 5 min, why not use crontab? fs_cli -x "jsrun xx.js" 2009/12/3 Oscav : > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to write > to a db some global informations (Global Variables) about FS like every 5 > minutes. > > Thanks. > > > -- > View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rob4manhere at gmail.com Thu Dec 3 05:31:16 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 3 Dec 2009 07:31:16 -0600 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <26625147.post@talk.nabble.com> References: <26625147.post@talk.nabble.com> Message-ID: What about cron? Create a cron entry like: */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" But if you're just dumping global variables, you could easily retrieve them directly from fs_cli without running an app and process the output however you'd like: /usr/local/freeswitch/bin/fs_cli -x "global_getvar" On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to > write > to a db some global informations (Global Variables) about FS like every 5 > minutes. > > Thanks. > > > -- > View this message in context: > http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/427638e5/attachment-0002.html From mike at jerris.com Thu Dec 3 05:48:55 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:48:55 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26621005.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> Message-ID: <9A0C4C44-B0D5-44F2-9C85-89FA27A89300@jerris.com> First off, maybe this conversation is better suited to the dev list, and second off, the current setup of where we do timers, where we poll, polling frequency and architecture is the result of 4+ years of ongoing testing and optimization. We have tried all different methods throughout. Sometimes what we found to be most efficient is not what we thought at first would be, but testing showed otherwise. We have always optimized the general case as to if there are many calls, and no suggestion would be implemented that hurts this case. That being said, if you could really come up with a way for this to be more efficient in any case, without sacrificing performance int he other cases, you are able to prove this with extensive test results, and you are able to prove that it does not impact for example call quality in any of the hundreds of edge cases that have led us to the point we are now, then we may be interested in taking such a patch. Mike On Dec 2, 2009, at 11:58 PM, eaf wrote: > > As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, it > could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" overrides > that. > > Yeah, there is a global timestamp... It's easy to workaround that for RTP > who calls switch_micro_time_now()... But if somebody accesses > runtime.timestamp directly, it's gonna be tough to grep for that. If only > this was C++... > > I'll play around. Never liked polling too much. Never could've guessed that > polling could be so useful for scalability ;) My naive implementation > would've pulled timestamp via system calls and would've done sleeping by > passing exact interval to select() instead of syncing with a pacing thread. > Which would be dead-quiet at idle time, but, of course, would stop scaling > at some point due to excessive number of system calls. > > Thanks. > > > Michael Jerris wrote: >> >> In short. No, you can not for many reasons. The milisecond tic is >> used throughout the code even when there is not any calls up. You can >> grep for switch_cond_next if you would like to see where but it is >> required to keep our global timestamp and for pacing the scheduler >> among other services that run all the time. >> >> Mike >> >> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >>> Can I reduce resolution of that timer thread 10 times? I mean, I >>> glanced >>> through the code, and see that among others (are there others?) RTP >>> and IVR >>> set up their timers that are subsequently managed by this thread. >>> RTP timers >>> should be eliminated by that setting you've suggested. IVR timers >>> are set at >>> 20ms... So, if the thread is set to wake up every 10ms instead of >>> 1ms it >>> should be able to wake up those IVR timers just fine. Right? >>> >>> That's a cool design to have one dedicated thread that maintains >>> accurate >>> timing and then broadcasts via condition variables to hundreds of >>> other >>> threads events that they can register for. I'm sure it's one of the >>> reasons >>> why FS scales so much better than Asterisk. But for poor low-end >>> setups that >>> sit in the closet, eat only 6W of power and hardly ever run more >>> than two >>> calls at the same time, can I hack it somehow to be more UNIX- >>> friendly? I.e. >>> make it stuck in select() or recv() when there is nothing to do, call >>> clock_gettime() right from the thread that wants and when it wants >>> to know >>> current time? >>> >>> Say, what if that thread is made to suspend on a condition variable >>> in case >>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>> thread comes up and adds its timer into the matrix, it could wake up >>> the >>> timer thread and enjoy accurate timing as needed, on demand? And in- >>> between >>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>> sitting on a wait queue in the kernel is way better than go back and >>> forth >>> incrementing counters that nobody even needs at the moment? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> idle is a 4 letter word to a realtime application. >>>> >>>> The core keeps a single high-priority thread to keep 1ms timing and >>>> expands >>>> that broadcasting >>>> to hundreds or thousand of threads who need accurate timing. >>>> >>>> Your choppy audio is caused by linksys lying about the packet len >>>> that >>>> it's >>>> using and we set our timer >>>> to the wrong speed. >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26621005.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Dec 3 05:50:19 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:50:19 -0500 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: References: <26625147.post@talk.nabble.com> Message-ID: <53E931F0-43C6-4756-B636-F17D7E076A1B@jerris.com> You could also use the scheduler to run the jsrun command inside FreeSWITCH. Mike On Dec 3, 2009, at 8:31 AM, Rob Forman wrote: > What about cron? > > Create a cron entry like: > */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" > > But if you're just dumping global variables, you could easily retrieve them directly from fs_cli without running an app and process the output however you'd like: > > /usr/local/freeswitch/bin/fs_cli -x "global_getvar" > > > On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > > Hi, > > Someone knows how to run periodically a JS script ?? The purpose is to write > to a db some global informations (Global Variables) about FS like every 5 > minutes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/70f1c40f/attachment-0002.html From mike at jerris.com Thu Dec 3 05:57:02 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 08:57:02 -0500 Subject: [Freeswitch-users] Best way to run originate calls through dial plan In-Reply-To: <26613841.post@talk.nabble.com> References: <26610094.post@talk.nabble.com> <87f2f3b90912020947v17b0b11fjfa06ced3d2879e5c@mail.gmail.com> <26613841.post@talk.nabble.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#originate Usage: originate |&() [] [] [] [] [] You can do this via shelling out to fs_cli like your example below or using esl directly from php: http://wiki.freeswitch.org/wiki/Esl Mike On Dec 2, 2009, at 1:23 PM, eaf wrote: > > I need a way to start a call from the PHP script to the originating number, > tell the party on that number to hold on, start another call to destination > number, and bridge everything together. On both legs I need to pass custom > caller ID. I can of course open direct connections to VOIP gateways right > from PHP, but I want to reuse existing routing rules in the dial plan, hence > I want to know what's the best way of making originate go through a specific > context of the dial plan. > > As for the number of calls per second, it's going to be only occasionally > used. > > > mercutioviz wrote: >> >> On Wed, Dec 2, 2009 at 6:47 AM, eaf wrote: >> >>> >>> What would be the best way of making originate() run call through a dial >>> plan >>> (compared to directly going to a specified VOIP gateway). Would it be >>> loopbacks, i.e. smth like this? >>> >>> /opt/freeswitch/bin/fs_cli -x "originate >>> >>> {ignore_early_media=true,origination_caller_id_number=xxxxxxxxxx}loopback/yyyyyyyyyy/default/XML >>> '&javascript(/opt/freeswitch/conf/dialplan/public/webcall.js >>> zzzzzzzzzz)'" >>> >>> The idea of this is that originate() sets up the first call, then >>> webcall.js >>> plays back a WAV, and bridges the first call with the second one (also >>> set >>> up via loopback). >>> >>> >> Could you describe the problem that you're trying to solve? That would >> make >> it easier to know if what you've come up with is the best solution. How >> many >> calls per second were you wanting to generate with this setup? >> -MC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Best-way-to-run-originate-calls-through-dial-plan-tp26610094p26613841.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Dec 3 06:08:04 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 09:08:04 -0500 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03c401ca73bf$1cea8600$56bf9200$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: The behavior is probably expected, the unhelpful error is probably undesirable but it would make a mess of the dial-plan to clean that up. Mike On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote: > Is this reasonable given it was the only call in FreeSwitch at the time? How > can this situation be corrected in the future? > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Wednesday, December 02, 2009 3:35 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Eavesdrop error? > > it probably just means the uuid was not retrieved from the db when you > called the eavesdrop exten which does the lookup on the uuid for the hash > key based on what ext you hit to retrieve the most recent uuid that called > that ext. > > > On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > Sorry, svn 15753 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb > Sent: Wednesday, December 02, 2009 2:08 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Eavesdrop error? > > I tried to use eavesdrop today and it did not work. The error message in the > log is: > > [ERR] mod_dptools.c:334 Usage: [all | ] > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > incorrect? > > http://pastebin.freeswitch.org/11363 > > Thanks Lars > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From erandr-junk at usa.net Thu Dec 3 06:17:54 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 06:17:54 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26621005.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> Message-ID: <26626634.post@talk.nabble.com> Oh, it's not just one timer thread... Why, why is sql_thread keeps on checking for messages every millisecond? Couldn't there be some signalling implemented that will make the thread suspend on condition variable or a socket/pipe in between? #0 do_sleep (t=1000) at src/switch_time.c:109 #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at src/switch_core_sqldb.c:783 Why does this sofia_profile_worker_thread keeps on looping checking for the queue? Have a semaphore! #0 do_sleep (t=1000) at src/switch_time.c:109 #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, obj=0x80f2490) at sofia.c:978 Nothing's happening on the box, but there are three threads that pretend to be actively busy with smth. Others at least sleep for hundreds of milliseconds, not for one. And there is even infrastructure present to do blocking pops: i.e. why couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed with 1ms sleeps? This looping is such a waste... eaf wrote: > > As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, > it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > overrides that. > > Yeah, there is a global timestamp... It's easy to workaround that for RTP > who calls switch_micro_time_now()... But if somebody accesses > runtime.timestamp directly, it's gonna be tough to grep for that. If only > this was C++... > > I'll play around. Never liked polling too much. Never could've guessed > that polling could be so useful for scalability ;) My naive implementation > would've pulled timestamp via system calls and would've done sleeping by > passing exact interval to select() instead of syncing with a pacing > thread. Which would be dead-quiet at idle time, but, of course, would stop > scaling at some point due to excessive number of system calls. > > Thanks. > > > Michael Jerris wrote: >> >> In short. No, you can not for many reasons. The milisecond tic is >> used throughout the code even when there is not any calls up. You can >> grep for switch_cond_next if you would like to see where but it is >> required to keep our global timestamp and for pacing the scheduler >> among other services that run all the time. >> >> Mike >> >> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >>> Can I reduce resolution of that timer thread 10 times? I mean, I >>> glanced >>> through the code, and see that among others (are there others?) RTP >>> and IVR >>> set up their timers that are subsequently managed by this thread. >>> RTP timers >>> should be eliminated by that setting you've suggested. IVR timers >>> are set at >>> 20ms... So, if the thread is set to wake up every 10ms instead of >>> 1ms it >>> should be able to wake up those IVR timers just fine. Right? >>> >>> That's a cool design to have one dedicated thread that maintains >>> accurate >>> timing and then broadcasts via condition variables to hundreds of >>> other >>> threads events that they can register for. I'm sure it's one of the >>> reasons >>> why FS scales so much better than Asterisk. But for poor low-end >>> setups that >>> sit in the closet, eat only 6W of power and hardly ever run more >>> than two >>> calls at the same time, can I hack it somehow to be more UNIX- >>> friendly? I.e. >>> make it stuck in select() or recv() when there is nothing to do, call >>> clock_gettime() right from the thread that wants and when it wants >>> to know >>> current time? >>> >>> Say, what if that thread is made to suspend on a condition variable >>> in case >>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>> thread comes up and adds its timer into the matrix, it could wake up >>> the >>> timer thread and enjoy accurate timing as needed, on demand? And in- >>> between >>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>> sitting on a wait queue in the kernel is way better than go back and >>> forth >>> incrementing counters that nobody even needs at the moment? >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> idle is a 4 letter word to a realtime application. >>>> >>>> The core keeps a single high-priority thread to keep 1ms timing and >>>> expands >>>> that broadcasting >>>> to hundreds or thousand of threads who need accurate timing. >>>> >>>> Your choppy audio is caused by linksys lying about the packet len >>>> that >>>> it's >>>> using and we set our timer >>>> to the wrong speed. >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26626634.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From abeka at greatiam.com Thu Dec 3 06:22:38 2009 From: abeka at greatiam.com (Otis) Date: Thu, 03 Dec 2009 14:22:38 +0000 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> Message-ID: <4B17C9AE.2010408@greatiam.com> Michael Collins wrote: > > > On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle > wrote: > > On Wed, Dec 02, Otis wrote: > Snip... > > > Thanks. > > > > I would like all extensions on say server A to be contactable > by those > > on server B and vice versa. > > The example I gave you should get you started. Let us know how > you get along. Have a read through the wiki pages like > > http://wiki.freeswitch.org/wiki/Dialplan_XML > http://wiki.freeswitch.org/wiki/Mod_dptools#Applications > http://wiki.freeswitch.org/wiki/Sofia > > --FC > > > Remember, too, that gateways are useful for doing auth/reg so having a > gateway on each box that registers to the other box is pretty handy. > If you run into any trouble trying to set it up you can ask here or > join us in #freeswitch on irc.freenode.net . > -MC Hi FC I used your code : replacing with my box's ip address. I have received any errors in the fs_cli console neither is there any reference to my box'x ipddress. Any way to check all is well ? And how do I join join us in #freeswitch on irc.freenode.net . ? Went to the freenode.net site and got lost. Will persevere. Thanks From william.suffill at gmail.com Thu Dec 3 06:34:44 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 3 Dec 2009 09:34:44 -0500 Subject: [Freeswitch-users] Bridging/Connecting Freeswitch servers In-Reply-To: <4B17C9AE.2010408@greatiam.com> References: <4B162271.1010306@greatiam.com> <20091202140155.GS31924@base.carmickle.com> <4B16A3F6.6000702@greatiam.com> <20091202175853.GW31924@base.carmickle.com> <87f2f3b90912021013j33764a46t936ab2a9bddb023e@mail.gmail.com> <4B17C9AE.2010408@greatiam.com> Message-ID: <6b65470d0912030634s4c9a304fu52ded127f98760a@mail.gmail.com> http://www.freeswitch.org/ On the right side. Join IRC Just fill in a nickname and click JOIN IRC -- W From erandr-junk at usa.net Thu Dec 3 06:55:21 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 06:55:21 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26626634.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> Message-ID: <26627246.post@talk.nabble.com> Btw, I have these popping up in my logs from time to time: 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration Detected! Syncing Clock In this case an incoming call rang to both FS and Asterisk, Asterisk picked up, but the surge of activity made FS timer thread miss a beat or two. eaf wrote: > > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > checking for messages every millisecond? Couldn't there be some signalling > implemented that will make the thread suspend on condition variable or a > socket/pipe in between? > > #0 do_sleep (t=1000) at src/switch_time.c:109 > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at > src/switch_core_sqldb.c:783 > > Why does this sofia_profile_worker_thread keeps on looping checking for > the queue? Have a semaphore! > > #0 do_sleep (t=1000) at src/switch_time.c:109 > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > obj=0x80f2490) at sofia.c:978 > > Nothing's happening on the box, but there are three threads that pretend > to be actively busy with smth. Others at least sleep for hundreds of > milliseconds, not for one. > > And there is even infrastructure present to do blocking pops: i.e. why > couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed > with 1ms sleeps? This looping is such a waste... > > > eaf wrote: >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> overrides that. >> >> Yeah, there is a global timestamp... It's easy to workaround that for RTP >> who calls switch_micro_time_now()... But if somebody accesses >> runtime.timestamp directly, it's gonna be tough to grep for that. If only >> this was C++... >> >> I'll play around. Never liked polling too much. Never could've guessed >> that polling could be so useful for scalability ;) My naive >> implementation would've pulled timestamp via system calls and would've >> done sleeping by passing exact interval to select() instead of syncing >> with a pacing thread. Which would be dead-quiet at idle time, but, of >> course, would stop scaling at some point due to excessive number of >> system calls. >> >> Thanks. >> >> >> Michael Jerris wrote: >>> >>> In short. No, you can not for many reasons. The milisecond tic is >>> used throughout the code even when there is not any calls up. You can >>> grep for switch_cond_next if you would like to see where but it is >>> required to keep our global timestamp and for pacing the scheduler >>> among other services that run all the time. >>> >>> Mike >>> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >>> >>>> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >>>> glanced >>>> through the code, and see that among others (are there others?) RTP >>>> and IVR >>>> set up their timers that are subsequently managed by this thread. >>>> RTP timers >>>> should be eliminated by that setting you've suggested. IVR timers >>>> are set at >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >>>> 1ms it >>>> should be able to wake up those IVR timers just fine. Right? >>>> >>>> That's a cool design to have one dedicated thread that maintains >>>> accurate >>>> timing and then broadcasts via condition variables to hundreds of >>>> other >>>> threads events that they can register for. I'm sure it's one of the >>>> reasons >>>> why FS scales so much better than Asterisk. But for poor low-end >>>> setups that >>>> sit in the closet, eat only 6W of power and hardly ever run more >>>> than two >>>> calls at the same time, can I hack it somehow to be more UNIX- >>>> friendly? I.e. >>>> make it stuck in select() or recv() when there is nothing to do, call >>>> clock_gettime() right from the thread that wants and when it wants >>>> to know >>>> current time? >>>> >>>> Say, what if that thread is made to suspend on a condition variable >>>> in case >>>> if there are no timers registered in TIMER_MATRIX? Then, if some other >>>> thread comes up and adds its timer into the matrix, it could wake up >>>> the >>>> timer thread and enjoy accurate timing as needed, on demand? And in- >>>> between >>>> the calls, when there is no RTP or IVR, it will all go silent? I mean, >>>> sitting on a wait queue in the kernel is way better than go back and >>>> forth >>>> incrementing counters that nobody even needs at the moment? >>>> >>>> >>>> Anthony Minessale-2 wrote: >>>>> >>>>> idle is a 4 letter word to a realtime application. >>>>> >>>>> The core keeps a single high-priority thread to keep 1ms timing and >>>>> expands >>>>> that broadcasting >>>>> to hundreds or thousand of threads who need accurate timing. >>>>> >>>>> Your choppy audio is caused by linksys lying about the packet len >>>>> that >>>>> it's >>>>> using and we set our timer >>>>> to the wrong speed. >>>>> >>>>> >>>> >>>> -- >>>> View this message in context: >>>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>> users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rupa at rupa.com Thu Dec 3 07:00:01 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 3 Dec 2009 09:00:01 -0600 Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> References: <26625147.post@talk.nabble.com> <23f91030912030528l11fc372ewde03b645af351c44@mail.gmail.com> Message-ID: If doing this, I'd suggest checking for a global var to see if the script should terminate itself. Otherwise, you'll have to bring down the whole freeswitch to stop this script. On Thu, Dec 3, 2009 at 7:28 AM, Seven Du wrote: > Not sure about js, but in lua, you can use luarun to run a > long-running script like > > > loop > do sth. > sleep 5min > end > > and also it can be set to start with freeswitch in lua.conf.xml > > I guess you can also use jsrun to run js. > > And, if you run every 5 min, why not use crontab? > > fs_cli -x "jsrun xx.js" > > > 2009/12/3 Oscav : > > > > Hi, > > > > Someone knows how to run periodically a JS script ?? The purpose is to > write > > to a db some global informations (Global Variables) about FS like every 5 > > minutes. > > > > Thanks. > > > > > > -- > > View this message in context: > http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/950f5139/attachment-0002.html From testeador01 at gmail.com Thu Dec 3 07:30:48 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 3 Dec 2009 10:30:48 -0500 Subject: [Freeswitch-users] Call transfer got broken for me Message-ID: Hello, It was all ok until yesterday when i updated to svn 15761(last update before that was about 4 days ago), Now I have this issue: someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext 200 200 picks up, then 200 transfers the call to 205 call gets lost (it used to transfer normal until the moment I updated) Today I updated to 15771 and the issue is still there. Can anyone help me figure out what is going on? Call log: http://pastebin.freeswitch.org/11374 thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/15d36757/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 07:56:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 09:56:14 -0600 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111420g794f6a79xe9fd1718285cfd33@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> Message-ID: <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> Try trunk again On Wed, Dec 2, 2009 at 5:33 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I am not sure what you are sending over the socket but you have a queued > hangup being processed on line 640 of your pastebin > are you executing any commands with a ! character in it by any chance or > executing the hangup app on purpose? > > > > > On Wed, Dec 2, 2009 at 2:16 PM, Kristian Kielhofner < > kristian.kielhofner at gmail.com> wrote: > >> Tony, >> >> Thanks for that but now it appears that the call just gets hung up >> on when the caller takes the callee off hold. Debug here: >> >> http://pastebin.freeswitch.org/11359 >> >> Thanks again! >> >> On Wed, Dec 2, 2009 at 1:13 PM, Anthony Minessale >> wrote: >> > I decided to just change the code so its more elegant to handle >> recursive >> > broadcasting so you can try again and see if that helps. >> > >> > >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/fbf05b86/attachment-0002.html From mike at jerris.com Thu Dec 3 07:59:44 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 10:59:44 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? Mike On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote: > I have read some of the archived emails about HA, loadbalancing, > failover etc and I am still a bit confused about how I could set up > some sort of resiliency with freeswitch. > > My situation is much less complex than the scenarios people were > talking about and I hoping the solution is similarly much less > complex. > > I have two machines. Both will run freeswitch and also an IVR > application with local databases. I will take care of the database, > application and configuration synchronization between the two > machines. Ideally the calls would be load balanced between the > machines and if any application falls down then the calls should go to > the other machine. Same if I take a machine down for whatever reason. > > If a machine goes down I am willing to "lose" those people who were > making a call at the time. I do have a flag in the application which > will stop answering the calls while processing the existing calls for > a graceful shutdown and hopefully the load balancer would shuttle the > calls to the other machine while this is happening. > > At this stage everything is done via SIP. > > My questions are... > > Do I have to have a sip proxy? If the answer is yes it seems like I > have to set up two sip proxies so I don't have another single point of > failure. Can I load the sip proxies on the same machine? Do I need two > more machines? > > If I take load balancing out of the picture would it be possible to do > a simple linux HA or a windows built in ip failover solution? Would a > simple IP failover work over UDP or would I have to use IAX and tcp/ip > ? > > Is it better to go the virtualization route? > > Sorry if these are dumb questions. I am just trying to get my head > wrapped around this. I don't need five nines (although that would be > awesome), I just want a reasonable degree of assurance that my app can > keep taking calls in case something weird happens. From mike at jerris.com Thu Dec 3 08:01:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:01:34 -0500 Subject: [Freeswitch-users] can't register Inphonex In-Reply-To: <005a01ca73c9$bc2dcf60$34896e20$@ca> References: <005a01ca73c9$bc2dcf60$34896e20$@ca> Message-ID: <30C82C4E-00FD-45BE-9D45-93FD0F99694E@jerris.com> You can turn up the full freeswitch debug or enable the siptrace on the sip profile to get more information about this. This looks like a nat related issue getting no response from the provider. A sip trace is probably the best tool to figure this one out. sofia profile internal siptrace on Mike On Dec 2, 2009, at 10:35 PM, John Lalande wrote: > I am new to FS having ditched Asterisk a few weeks ago. I have iptel registered but I cannot get Inphonex to work. I am using the settings fromhttp://wiki.freeswitch.org/wiki/Provider_Configuration:_Inphonex to no avail. > > The error displayed in the console is "2009-12-02 21:32:55.243917 [ERR] sofia_reg.c:1442 inphonex Registration Failed with status Request Timeout [408]." > > Is there some way to debug this? sofia status displays: > > Name Type Data State > ================================================================================================= > external profile sip:mod_sofia at 192.168.125.15:5080 RUNNING (0) > example.com gateway sip:joeuser at example.com NOREG > inphonex gateway sip:5285418 at sip.inphonex.com FAILED (retry: 28s) > iptel gateway sip:jlalande at sip.iptel.org REGED > internal profile sip:mod_sofia at 192.168.125.15:5060 RUNNING (0) > internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) > 192.168.125.15 alias internal ALIASED > ================================================================================================= > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/937cd8cb/attachment-0002.html From mike at jerris.com Thu Dec 3 08:07:34 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:07:34 -0500 Subject: [Freeswitch-users] Gateway issue with no audio In-Reply-To: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> References: <59ad9ca10912030136n6b79cc83xb9f8608c0575fd9b@mail.gmail.com> Message-ID: You may want to try this again with latest svn trunk. We have done quite a lot of work to make nat support much better sense 1.0.4 Mike p.s. I can't comment about version 1.4 due to broken flux capacitor. On Dec 3, 2009, at 4:36 AM, Henry Huang wrote: > My freeswitch is using public IP. I setup a gateway registering to voipstunt, and put it under internal profile. I tried to make call, and I got no RTP back from the provider... Tried treating NAT issue by changing IP address, internal IP, external IP. But no use, still getting no audio. > > Finally, I gave up play around with the internal profile and put the gateway settings under external profile. And magically, it worked. I am getting audio now. But it leads me to wonders, what's the core difference between external profile and internal profile. Even if I set the external SIP IP and exteranl RTP IP to the public IP in internal profile, I am still getting no audio. Can anyone clear the concept for me here? > > by the way, I am using freeswitch 1.4 stable version. > > > > -- > Henry Huang > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/aaabc8cd/attachment-0002.html From mike at jerris.com Thu Dec 3 08:10:59 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:10:59 -0500 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <26625105.post@talk.nabble.com> References: <26625105.post@talk.nabble.com> Message-ID: <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> with the right clients, it nearly always works well. with a client that does not support stun or at least rfc 3581 the results are much more sketchy and require more hacks on the server side, but with enough effort can almost always be made to work. Mike On Dec 3, 2009, at 7:17 AM, Fred-145 wrote: > > Hello > > In a thread back in March, I read that support for IAX in FreeSwitch is a > bit of kludge and since there's not much demand for it, chances are it won't > improve in the foreseeable future. > > So I'd like some feedback from users who routinely connect to a FreeSwitch > server from various venues, ie. wifi hotspots at McD, Ethernet LAN in > hotels, etc. (in my case, the FreeSwitch server is located in a private > network behind a NAT router with SIP/RTP ports statically mapped.) > > Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) > ports fail being opened dynamically to work properly, or does SIP today > really work well over NAT firewalls? > > Thank you. > -- > View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26625105.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 3 08:16:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:16:26 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26627246.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> Message-ID: <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> If you see that message then your machine/os/combo is having some problems keeping up. It's not the timer missing anything its the monotonic clock detecting a 1 second or more differential from what its next prediction for the time should be. The best way to trigger this would be to suspend FS with control-z or attach to it with gdb blocking the entire process, that 1ms thread would have to miss 1000 iterations to trigger that warning. Btw, that error message is at line 471 not 473 so you are using modified code. Its possible your box has a bad monotonic timer, you can set under in switch.conf.xml We are now starting to guess you are using some small embedded type platform perhaps? I've run FS even on a nokia n810 and never caused that message to fire. if 1 call can interrupt the cpu enough to cause noticeable issues you might want to consider running the process at a greater priority by using the -hp command line arg or at least nice it Why don't you tell us the whole story about what OS/platform you are using here rather that form conjectures about what is wrong with our code that thousands of people are happy with. On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > > Btw, I have these popping up in my logs from time to time: > > 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > Detected! Syncing Clock > > In this case an incoming call rang to both FS and Asterisk, Asterisk picked > up, but the surge of activity made FS timer thread miss a beat or two. > > > eaf wrote: > > > > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > > checking for messages every millisecond? Couldn't there be some > signalling > > implemented that will make the thread suspend on condition variable or a > > socket/pipe in between? > > > > #0 do_sleep (t=1000) at src/switch_time.c:109 > > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) at > > src/switch_core_sqldb.c:783 > > > > Why does this sofia_profile_worker_thread keeps on looping checking for > > the queue? Have a semaphore! > > > > #0 do_sleep (t=1000) at src/switch_time.c:109 > > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > > obj=0x80f2490) at sofia.c:978 > > > > Nothing's happening on the box, but there are three threads that pretend > > to be actively busy with smth. Others at least sleep for hundreds of > > milliseconds, not for one. > > > > And there is even infrastructure present to do blocking pops: i.e. why > > couldn't sqldb thread do queue_pop() instead of queue_trypop() intermixed > > with 1ms sleeps? This looping is such a waste... > > > > > > eaf wrote: > >> > >> As I see it, switch_cond_next() currently is just a do_sleep(1000). Yes, > >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> overrides that. > >> > >> Yeah, there is a global timestamp... It's easy to workaround that for > RTP > >> who calls switch_micro_time_now()... But if somebody accesses > >> runtime.timestamp directly, it's gonna be tough to grep for that. If > only > >> this was C++... > >> > >> I'll play around. Never liked polling too much. Never could've guessed > >> that polling could be so useful for scalability ;) My naive > >> implementation would've pulled timestamp via system calls and would've > >> done sleeping by passing exact interval to select() instead of syncing > >> with a pacing thread. Which would be dead-quiet at idle time, but, of > >> course, would stop scaling at some point due to excessive number of > >> system calls. > >> > >> Thanks. > >> > >> > >> Michael Jerris wrote: > >>> > >>> In short. No, you can not for many reasons. The milisecond tic is > >>> used throughout the code even when there is not any calls up. You can > >>> grep for switch_cond_next if you would like to see where but it is > >>> required to keep our global timestamp and for pacing the scheduler > >>> among other services that run all the time. > >>> > >>> Mike > >>> > >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >>> > >>>> > >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >>>> glanced > >>>> through the code, and see that among others (are there others?) RTP > >>>> and IVR > >>>> set up their timers that are subsequently managed by this thread. > >>>> RTP timers > >>>> should be eliminated by that setting you've suggested. IVR timers > >>>> are set at > >>>> 20ms... So, if the thread is set to wake up every 10ms instead of > >>>> 1ms it > >>>> should be able to wake up those IVR timers just fine. Right? > >>>> > >>>> That's a cool design to have one dedicated thread that maintains > >>>> accurate > >>>> timing and then broadcasts via condition variables to hundreds of > >>>> other > >>>> threads events that they can register for. I'm sure it's one of the > >>>> reasons > >>>> why FS scales so much better than Asterisk. But for poor low-end > >>>> setups that > >>>> sit in the closet, eat only 6W of power and hardly ever run more > >>>> than two > >>>> calls at the same time, can I hack it somehow to be more UNIX- > >>>> friendly? I.e. > >>>> make it stuck in select() or recv() when there is nothing to do, call > >>>> clock_gettime() right from the thread that wants and when it wants > >>>> to know > >>>> current time? > >>>> > >>>> Say, what if that thread is made to suspend on a condition variable > >>>> in case > >>>> if there are no timers registered in TIMER_MATRIX? Then, if some other > >>>> thread comes up and adds its timer into the matrix, it could wake up > >>>> the > >>>> timer thread and enjoy accurate timing as needed, on demand? And in- > >>>> between > >>>> the calls, when there is no RTP or IVR, it will all go silent? I mean, > >>>> sitting on a wait queue in the kernel is way better than go back and > >>>> forth > >>>> incrementing counters that nobody even needs at the moment? > >>>> > >>>> > >>>> Anthony Minessale-2 wrote: > >>>>> > >>>>> idle is a 4 letter word to a realtime application. > >>>>> > >>>>> The core keeps a single high-priority thread to keep 1ms timing and > >>>>> expands > >>>>> that broadcasting > >>>>> to hundreds or thousand of threads who need accurate timing. > >>>>> > >>>>> Your choppy audio is caused by linksys lying about the packet len > >>>>> that > >>>>> it's > >>>>> using and we set our timer > >>>>> to the wrong speed. > >>>>> > >>>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>>> users > >>>> http://www.freeswitch.org > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/061f041e/attachment-0002.html From mike at jerris.com Thu Dec 3 08:21:43 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 11:21:43 -0500 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: References: Message-ID: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> what revision were you at prior to upgrade or can you narrow the range of versions that broke this any more (or even better the exact version that broke this). Please post this bug to http://jira.freeswitch.org. Mike On Dec 3, 2009, at 10:30 AM, Milena wrote: > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/5e7c3a92/attachment-0002.html From testeador01 at gmail.com Thu Dec 3 08:29:36 2009 From: testeador01 at gmail.com (Milena) Date: Thu, 3 Dec 2009 11:29:36 -0500 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> References: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> Message-ID: This got fixed in version 15773, thank you very much 2009/12/3 Michael Jerris > what revision were you at prior to upgrade or can you narrow the range of > versions that broke this any more (or even better the exact version that > broke this). Please post this bug to http://jira.freeswitch.org. > > Mike > > On Dec 3, 2009, at 10:30 AM, Milena wrote: > > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update > before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext > 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/f5254fa5/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 08:32:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:32:19 -0600 Subject: [Freeswitch-users] Call transfer got broken for me In-Reply-To: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> References: <5418B57F-A2E2-417D-9EB5-681069D0031D@jerris.com> Message-ID: <191c3a030912030832o4af484afrad086b848d0c6384@mail.gmail.com> to late it's fixed now. On Thu, Dec 3, 2009 at 10:21 AM, Michael Jerris wrote: > what revision were you at prior to upgrade or can you narrow the range of > versions that broke this any more (or even better the exact version that > broke this). Please post this bug to http://jira.freeswitch.org. > > Mike > > On Dec 3, 2009, at 10:30 AM, Milena wrote: > > Hello, > > It was all ok until yesterday when i updated to svn 15761(last update > before that was about 4 days ago), Now I have this issue: > > someone from the pstn (5555555) calls through my FXO gw (10.1.1.90) to ext > 200 > 200 picks up, then 200 transfers the call to 205 > call gets lost (it used to transfer normal until the moment I updated) > > Today I updated to 15771 and the issue is still there. > Can anyone help me figure out what is going on? > > Call log: http://pastebin.freeswitch.org/11374 > > thank you > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/368a0e12/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 08:35:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 10:35:56 -0600 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: <191c3a030912030835s3884a2fal73cf9527041e023b@mail.gmail.com> you could check if the uuid is blank with an expression and playback an audio warning that it's an invalid call. On Thu, Dec 3, 2009 at 8:08 AM, Michael Jerris wrote: > The behavior is probably expected, the unhelpful error is probably > undesirable but it would make a mess of the dial-plan to clean that up. > > Mike > > On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote: > > > Is this reasonable given it was the only call in FreeSwitch at the time? > How > > can this situation be corrected in the future? > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: Wednesday, December 02, 2009 3:35 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Eavesdrop error? > > > > it probably just means the uuid was not retrieved from the db when you > > called the eavesdrop exten which does the lookup on the uuid for the hash > > key based on what ext you hit to retrieve the most recent uuid that > called > > that ext. > > > > > > On Wed, Dec 2, 2009 at 5:22 PM, Lars Zeb wrote: > > Sorry, svn 15753 > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars > Zeb > > Sent: Wednesday, December 02, 2009 2:08 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Eavesdrop error? > > > > I tried to use eavesdrop today and it did not work. The error message in > the > > log is: > > > > [ERR] mod_dptools.c:334 Usage: [all | ] > > > > I simply dialed 881010, trying to eavesdrop on extension 1010. Is this > > incorrect? > > > > http://pastebin.freeswitch.org/11363 > > > > Thanks Lars > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/44620e04/attachment-0002.html From lists at redbonez.net Thu Dec 3 08:56:23 2009 From: lists at redbonez.net (Adam Ford) Date: Thu, 3 Dec 2009 09:56:23 -0700 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <012801ca7439$8cc10ba0$a64322e0$@net> Have you checked out Redfone? While I haven't attempted to implement it yet, my Redfone foneBridge2 claims to be able to handle load balancing and failover between two Asterisk/Freeswitch servers. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Uckun Sent: Wednesday, December 02, 2009 5:50 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] HA questions. I have read some of the archived emails about HA, loadbalancing, failover etc and I am still a bit confused about how I could set up some sort of resiliency with freeswitch. My situation is much less complex than the scenarios people were talking about and I hoping the solution is similarly much less complex. I have two machines. Both will run freeswitch and also an IVR application with local databases. I will take care of the database, application and configuration synchronization between the two machines. Ideally the calls would be load balanced between the machines and if any application falls down then the calls should go to the other machine. Same if I take a machine down for whatever reason. If a machine goes down I am willing to "lose" those people who were making a call at the time. I do have a flag in the application which will stop answering the calls while processing the existing calls for a graceful shutdown and hopefully the load balancer would shuttle the calls to the other machine while this is happening. At this stage everything is done via SIP. My questions are... Do I have to have a sip proxy? If the answer is yes it seems like I have to set up two sip proxies so I don't have another single point of failure. Can I load the sip proxies on the same machine? Do I need two more machines? If I take load balancing out of the picture would it be possible to do a simple linux HA or a windows built in ip failover solution? Would a simple IP failover work over UDP or would I have to use IAX and tcp/ip ? Is it better to go the virtualization route? Sorry if these are dumb questions. I am just trying to get my head wrapped around this. I don't need five nines (although that would be awesome), I just want a reasonable degree of assurance that my app can keep taking calls in case something weird happens. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From erandr-junk at usa.net Thu Dec 3 09:29:46 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 09:29:46 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> Message-ID: <26629856.post@talk.nabble.com> I'm sorry if I sounded that way. Did mean to. :) Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm Line offset difference is due to some minor logging changes I made to see who's allocating timers and how often. This way I found MOH streaming and that RTP still allocates timers even when it's set to none in the profile. I feel that this platform turned out to be underpowered for FS because it cannot meet its scheduling expectations. I guess, some degree of kernel tweaking or setting priorities will fix that. Meanwhile I just got rid of the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms thread in two (one blocked and waiting for new commands in the SQL queue, the other one checking registrations and gateways with 1sec interval), and don't know yet what to do about the timer thread. Again, I apologize for stupid or accusing questions, I'm just trying to see how FS can be made friendlier to this board. Or the board be made friendlier to FS ;) Anthony Minessale-2 wrote: > > If you see that message then your machine/os/combo is having some problems > keeping up. > It's not the timer missing anything its the monotonic clock detecting a 1 > second or more differential from what its next prediction for the time > should be. The best way to trigger this would be to suspend FS with > control-z or attach to it with gdb blocking the entire process, that 1ms > thread would have to miss 1000 iterations to trigger that warning. > > Btw, that error message is at line 471 not 473 so you are using modified > code. > > Its possible your box has a bad monotonic timer, you can set > > > > under in switch.conf.xml > > We are now starting to guess you are using some small embedded type > platform > perhaps? > I've run FS even on a nokia n810 and never caused that message to fire. > > if 1 call can interrupt the cpu enough to cause noticeable issues you > might > want to consider running the process at a > greater priority by using the -hp command line arg or at least nice it > > Why don't you tell us the whole story about what OS/platform you are using > here rather that form conjectures about what is wrong with our code that > thousands of people are happy with. > > > > > > > > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > >> >> Btw, I have these popping up in my logs from time to time: >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >> Detected! Syncing Clock >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk >> picked >> up, but the surge of activity made FS timer thread miss a beat or two. >> >> >> eaf wrote: >> > >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps on >> > checking for messages every millisecond? Couldn't there be some >> signalling >> > implemented that will make the thread suspend on condition variable or >> a >> > socket/pipe in between? >> > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) >> at >> > src/switch_core_sqldb.c:783 >> > >> > Why does this sofia_profile_worker_thread keeps on looping checking for >> > the queue? Have a semaphore! >> > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >> > obj=0x80f2490) at sofia.c:978 >> > >> > Nothing's happening on the box, but there are three threads that >> pretend >> > to be actively busy with smth. Others at least sleep for hundreds of >> > milliseconds, not for one. >> > >> > And there is even infrastructure present to do blocking pops: i.e. why >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() >> intermixed >> > with 1ms sleeps? This looping is such a waste... >> > >> > >> > eaf wrote: >> >> >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). >> Yes, >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> >> overrides that. >> >> >> >> Yeah, there is a global timestamp... It's easy to workaround that for >> RTP >> >> who calls switch_micro_time_now()... But if somebody accesses >> >> runtime.timestamp directly, it's gonna be tough to grep for that. If >> only >> >> this was C++... >> >> >> >> I'll play around. Never liked polling too much. Never could've guessed >> >> that polling could be so useful for scalability ;) My naive >> >> implementation would've pulled timestamp via system calls and would've >> >> done sleeping by passing exact interval to select() instead of syncing >> >> with a pacing thread. Which would be dead-quiet at idle time, but, of >> >> course, would stop scaling at some point due to excessive number of >> >> system calls. >> >> >> >> Thanks. >> >> >> >> >> >> Michael Jerris wrote: >> >>> >> >>> In short. No, you can not for many reasons. The milisecond tic is >> >>> used throughout the code even when there is not any calls up. You >> can >> >>> grep for switch_cond_next if you would like to see where but it is >> >>> required to keep our global timestamp and for pacing the scheduler >> >>> among other services that run all the time. >> >>> >> >>> Mike >> >>> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >>> >> >>>> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >> >>>> glanced >> >>>> through the code, and see that among others (are there others?) RTP >> >>>> and IVR >> >>>> set up their timers that are subsequently managed by this thread. >> >>>> RTP timers >> >>>> should be eliminated by that setting you've suggested. IVR timers >> >>>> are set at >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >> >>>> 1ms it >> >>>> should be able to wake up those IVR timers just fine. Right? >> >>>> >> >>>> That's a cool design to have one dedicated thread that maintains >> >>>> accurate >> >>>> timing and then broadcasts via condition variables to hundreds of >> >>>> other >> >>>> threads events that they can register for. I'm sure it's one of the >> >>>> reasons >> >>>> why FS scales so much better than Asterisk. But for poor low-end >> >>>> setups that >> >>>> sit in the closet, eat only 6W of power and hardly ever run more >> >>>> than two >> >>>> calls at the same time, can I hack it somehow to be more UNIX- >> >>>> friendly? I.e. >> >>>> make it stuck in select() or recv() when there is nothing to do, >> call >> >>>> clock_gettime() right from the thread that wants and when it wants >> >>>> to know >> >>>> current time? >> >>>> >> >>>> Say, what if that thread is made to suspend on a condition variable >> >>>> in case >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some >> other >> >>>> thread comes up and adds its timer into the matrix, it could wake up >> >>>> the >> >>>> timer thread and enjoy accurate timing as needed, on demand? And in- >> >>>> between >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I >> mean, >> >>>> sitting on a wait queue in the kernel is way better than go back and >> >>>> forth >> >>>> incrementing counters that nobody even needs at the moment? >> >>>> >> >>>> >> >>>> Anthony Minessale-2 wrote: >> >>>>> >> >>>>> idle is a 4 letter word to a realtime application. >> >>>>> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing and >> >>>>> expands >> >>>>> that broadcasting >> >>>>> to hundreds or thousand of threads who need accurate timing. >> >>>>> >> >>>>> Your choppy audio is caused by linksys lying about the packet len >> >>>>> that >> >>>>> it's >> >>>>> using and we set our timer >> >>>>> to the wrong speed. >> >>>>> >> >>>>> >> >>>> >> >>>> -- >> >>>> View this message in context: >> >>>> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> >>>> users >> >>>> http://www.freeswitch.org >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >>> >> >> >> >> >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristian.kielhofner at gmail.com Thu Dec 3 09:33:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 3 Dec 2009 12:33:50 -0500 Subject: [Freeswitch-users] [local_stream://moh] already broadcasting...broadcast aborted In-Reply-To: <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> References: <2d9149cd0911111319k3983e2f4oc2bf397269a44fe7@mail.gmail.com> <2d9149cd0911111433w6bc7d11bp6dc859647a22880d@mail.gmail.com> <191c3a030911111914u6628448bhcdf04a11ed472407@mail.gmail.com> <2d9149cd0911200841g8b2f884x4502428b1490e329@mail.gmail.com> <191c3a030911201934h547296c3jc248f28a31736494@mail.gmail.com> <2d9149cd0912020835n4bf12a12wd64773020a6ea782@mail.gmail.com> <191c3a030912021013n1c1380e1h5bcf19575b225f00@mail.gmail.com> <2d9149cd0912021216w14e74549x9bdc029a4697ec06@mail.gmail.com> <191c3a030912021533p514209baq42f4dcf078d29225@mail.gmail.com> <191c3a030912030756s7039bb77ld8ee8e85593bf777@mail.gmail.com> Message-ID: <2d9149cd0912030933k110a89e2j12a8d44bfcb86bbb@mail.gmail.com> Tony, The call no longer hangs up but we still only get hold music in one direction - if the callee places the caller on hold there is no music. PB here: http://pastebin.freeswitch.org/11378 This was on rev 15773. Thanks again Tony! On Thu, Dec 3, 2009 at 10:56 AM, Anthony Minessale wrote: > Try trunk again > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From kristian.kielhofner at gmail.com Thu Dec 3 09:44:27 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 3 Dec 2009 12:44:27 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: <2d9149cd0912030944g5a0b1145q1c292e9bd1e4a11d@mail.gmail.com> I don't think it's the board itself... We have extensively tested FreeSwitch (no modifications) on that exact board with AstLinux and have it running at multiple customer locations. No timing errors, no warnings or errors of any kind. Pretty standard really just don't expect too much from the LX800 (transcoding, resampling, massive numbers of calls, etc). On Thu, Dec 3, 2009 at 12:29 PM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made friendlier > to FS ;) > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From abeka at greatiam.com Thu Dec 3 09:46:07 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 17:46:07 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing Message-ID: <4B17F95F.2000108@greatiam.com> I have copied 1001.xml in directory/default to a test user 2319.xm changing or instances of 1001 in the file to 2319. I then went into default.xml in directory folder and in one of the groups just mimicked 1001 details by changing 1001 to 2319. Connecting to FS gives Forbidden message. However 1001 connects without a problem. What have I missed ? Is there a place that just puts things in do this and that and that to create a new user ? Thanks From mike at jerris.com Thu Dec 3 09:50:26 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 12:50:26 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: I know people with hardware out there in production based on arm11 and those are pretty small processors, not sure how they compare to this. In regards to the DISABLE_1MS_COND, try getting rid of that, it did increase performance on the high end but may be better for you on the low end with lower compute on idle busy loops. Mike On Dec 3, 2009, at 12:29 PM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made friendlier > to FS ;) > > > Anthony Minessale-2 wrote: >> >> If you see that message then your machine/os/combo is having some problems >> keeping up. >> It's not the timer missing anything its the monotonic clock detecting a 1 >> second or more differential from what its next prediction for the time >> should be. The best way to trigger this would be to suspend FS with >> control-z or attach to it with gdb blocking the entire process, that 1ms >> thread would have to miss 1000 iterations to trigger that warning. >> >> Btw, that error message is at line 471 not 473 so you are using modified >> code. >> >> Its possible your box has a bad monotonic timer, you can set >> >> >> >> under in switch.conf.xml >> >> We are now starting to guess you are using some small embedded type >> platform >> perhaps? >> I've run FS even on a nokia n810 and never caused that message to fire. >> >> if 1 call can interrupt the cpu enough to cause noticeable issues you >> might >> want to consider running the process at a >> greater priority by using the -hp command line arg or at least nice it >> >> Why don't you tell us the whole story about what OS/platform you are using >> here rather that form conjectures about what is wrong with our code that >> thousands of people are happy with. >> >> >> >> >> >> >> >> On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: >> >>> >>> Btw, I have these popping up in my logs from time to time: >>> >>> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >>> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >>> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >>> Detected! Syncing Clock >>> >>> In this case an incoming call rang to both FS and Asterisk, Asterisk >>> picked >>> up, but the surge of activity made FS timer thread miss a beat or two. >>> >>> >>> eaf wrote: >>>> >>>> Oh, it's not just one timer thread... Why, why is sql_thread keeps on >>>> checking for messages every millisecond? Couldn't there be some >>> signalling >>>> implemented that will make the thread suspend on condition variable or >>> a >>>> socket/pipe in between? >>>> >>>> #0 do_sleep (t=1000) at src/switch_time.c:109 >>>> #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) >>> at >>>> src/switch_core_sqldb.c:783 >>>> >>>> Why does this sofia_profile_worker_thread keeps on looping checking for >>>> the queue? Have a semaphore! >>>> >>>> #0 do_sleep (t=1000) at src/switch_time.c:109 >>>> #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >>>> obj=0x80f2490) at sofia.c:978 >>>> >>>> Nothing's happening on the box, but there are three threads that >>> pretend >>>> to be actively busy with smth. Others at least sleep for hundreds of >>>> milliseconds, not for one. >>>> >>>> And there is even infrastructure present to do blocking pops: i.e. why >>>> couldn't sqldb thread do queue_pop() instead of queue_trypop() >>> intermixed >>>> with 1ms sleeps? This looping is such a waste... >>>> >>>> >>>> eaf wrote: >>>>> >>>>> As I see it, switch_cond_next() currently is just a do_sleep(1000). >>> Yes, >>>>> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >>>>> overrides that. >>>>> >>>>> Yeah, there is a global timestamp... It's easy to workaround that for >>> RTP >>>>> who calls switch_micro_time_now()... But if somebody accesses >>>>> runtime.timestamp directly, it's gonna be tough to grep for that. If >>> only >>>>> this was C++... >>>>> >>>>> I'll play around. Never liked polling too much. Never could've guessed >>>>> that polling could be so useful for scalability ;) My naive >>>>> implementation would've pulled timestamp via system calls and would've >>>>> done sleeping by passing exact interval to select() instead of syncing >>>>> with a pacing thread. Which would be dead-quiet at idle time, but, of >>>>> course, would stop scaling at some point due to excessive number of >>>>> system calls. >>>>> >>>>> Thanks. >>>>> >>>>> >>>>> Michael Jerris wrote: >>>>>> >>>>>> In short. No, you can not for many reasons. The milisecond tic is >>>>>> used throughout the code even when there is not any calls up. You >>> can >>>>>> grep for switch_cond_next if you would like to see where but it is >>>>>> required to keep our global timestamp and for pacing the scheduler >>>>>> among other services that run all the time. >>>>>> >>>>>> Mike >>>>>> >>>>>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >>>>>> >>>>>>> >>>>>>> Can I reduce resolution of that timer thread 10 times? I mean, I >>>>>>> glanced >>>>>>> through the code, and see that among others (are there others?) RTP >>>>>>> and IVR >>>>>>> set up their timers that are subsequently managed by this thread. >>>>>>> RTP timers >>>>>>> should be eliminated by that setting you've suggested. IVR timers >>>>>>> are set at >>>>>>> 20ms... So, if the thread is set to wake up every 10ms instead of >>>>>>> 1ms it >>>>>>> should be able to wake up those IVR timers just fine. Right? >>>>>>> >>>>>>> That's a cool design to have one dedicated thread that maintains >>>>>>> accurate >>>>>>> timing and then broadcasts via condition variables to hundreds of >>>>>>> other >>>>>>> threads events that they can register for. I'm sure it's one of the >>>>>>> reasons >>>>>>> why FS scales so much better than Asterisk. But for poor low-end >>>>>>> setups that >>>>>>> sit in the closet, eat only 6W of power and hardly ever run more >>>>>>> than two >>>>>>> calls at the same time, can I hack it somehow to be more UNIX- >>>>>>> friendly? I.e. >>>>>>> make it stuck in select() or recv() when there is nothing to do, >>> call >>>>>>> clock_gettime() right from the thread that wants and when it wants >>>>>>> to know >>>>>>> current time? >>>>>>> >>>>>>> Say, what if that thread is made to suspend on a condition variable >>>>>>> in case >>>>>>> if there are no timers registered in TIMER_MATRIX? Then, if some >>> other >>>>>>> thread comes up and adds its timer into the matrix, it could wake up >>>>>>> the >>>>>>> timer thread and enjoy accurate timing as needed, on demand? And in- >>>>>>> between >>>>>>> the calls, when there is no RTP or IVR, it will all go silent? I >>> mean, >>>>>>> sitting on a wait queue in the kernel is way better than go back and >>>>>>> forth >>>>>>> incrementing counters that nobody even needs at the moment? >>>>>>> >>>>>>> >>>>>>> Anthony Minessale-2 wrote: >>>>>>>> >>>>>>>> idle is a 4 letter word to a realtime application. >>>>>>>> >>>>>>>> The core keeps a single high-priority thread to keep 1ms timing and >>>>>>>> expands >>>>>>>> that broadcasting >>>>>>>> to hundreds or thousand of threads who need accurate timing. >>>>>>>> >>>>>>>> Your choppy audio is caused by linksys lying about the packet len >>>>>>>> that >>>>>>>> it's >>>>>>>> using and we set our timer >>>>>>>> to the wrong speed. >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> -- >>>>>>> View this message in context: >>>>>>> >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >>>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>>>>>> users >>>>>>> http://www.freeswitch.org >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Dec 3 09:57:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 09:57:53 -0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B17F95F.2000108@greatiam.com> References: <4B17F95F.2000108@greatiam.com> Message-ID: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah wrote: > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > Did you execute "reloadxml" from the fs cli before trying to connect with 2319? Also I'm assuming that "2319.xm" is a typo and you actually created "2319.xml" in the default/directory subdir. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/27520206/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 10:10:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 12:10:05 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26629856.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> Message-ID: <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> What about the things I spent time suggesting in my last email? Did you try them because I was actually curious if they made any impact. On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > > I'm sorry if I sounded that way. Did mean to. :) > > Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 > chip > and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > > Line offset difference is due to some minor logging changes I made to see > who's allocating timers and how often. This way I found MOH streaming and > that RTP still allocates timers even when it's set to none in the profile. > > I feel that this platform turned out to be underpowered for FS because it > cannot meet its scheduling expectations. I guess, some degree of kernel > tweaking or setting priorities will fix that. Meanwhile I just got rid of > the SQLDB 1ms thread via -nosql command line option, split sofia worker 1ms > thread in two (one blocked and waiting for new commands in the SQL queue, > the other one checking registrations and gateways with 1sec interval), and > don't know yet what to do about the timer thread. > > Again, I apologize for stupid or accusing questions, I'm just trying to see > how FS can be made friendlier to this board. Or the board be made > friendlier > to FS ;) > > > Anthony Minessale-2 wrote: > > > > If you see that message then your machine/os/combo is having some > problems > > keeping up. > > It's not the timer missing anything its the monotonic clock detecting a 1 > > second or more differential from what its next prediction for the time > > should be. The best way to trigger this would be to suspend FS with > > control-z or attach to it with gdb blocking the entire process, that 1ms > > thread would have to miss 1000 iterations to trigger that warning. > > > > Btw, that error message is at line 471 not 473 so you are using modified > > code. > > > > Its possible your box has a bad monotonic timer, you can set > > > > > > > > under in switch.conf.xml > > > > We are now starting to guess you are using some small embedded type > > platform > > perhaps? > > I've run FS even on a nokia n810 and never caused that message to fire. > > > > if 1 call can interrupt the cpu enough to cause noticeable issues you > > might > > want to consider running the process at a > > greater priority by using the -hp command line arg or at least nice it > > > > Why don't you tell us the whole story about what OS/platform you are > using > > here rather that form conjectures about what is wrong with our code that > > thousands of people are happy with. > > > > > > > > > > > > > > > > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > > > >> > >> Btw, I have these popping up in my logs from time to time: > >> > >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > >> Detected! Syncing Clock > >> > >> In this case an incoming call rang to both FS and Asterisk, Asterisk > >> picked > >> up, but the surge of activity made FS timer thread miss a beat or two. > >> > >> > >> eaf wrote: > >> > > >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps on > >> > checking for messages every millisecond? Couldn't there be some > >> signalling > >> > implemented that will make the thread suspend on condition variable or > >> a > >> > socket/pipe in between? > >> > > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, obj=0x0) > >> at > >> > src/switch_core_sqldb.c:783 > >> > > >> > Why does this sofia_profile_worker_thread keeps on looping checking > for > >> > the queue? Have a semaphore! > >> > > >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, > >> > obj=0x80f2490) at sofia.c:978 > >> > > >> > Nothing's happening on the box, but there are three threads that > >> pretend > >> > to be actively busy with smth. Others at least sleep for hundreds of > >> > milliseconds, not for one. > >> > > >> > And there is even infrastructure present to do blocking pops: i.e. why > >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() > >> intermixed > >> > with 1ms sleeps? This looping is such a waste... > >> > > >> > > >> > eaf wrote: > >> >> > >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). > >> Yes, > >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> >> overrides that. > >> >> > >> >> Yeah, there is a global timestamp... It's easy to workaround that for > >> RTP > >> >> who calls switch_micro_time_now()... But if somebody accesses > >> >> runtime.timestamp directly, it's gonna be tough to grep for that. If > >> only > >> >> this was C++... > >> >> > >> >> I'll play around. Never liked polling too much. Never could've > guessed > >> >> that polling could be so useful for scalability ;) My naive > >> >> implementation would've pulled timestamp via system calls and > would've > >> >> done sleeping by passing exact interval to select() instead of > syncing > >> >> with a pacing thread. Which would be dead-quiet at idle time, but, of > >> >> course, would stop scaling at some point due to excessive number of > >> >> system calls. > >> >> > >> >> Thanks. > >> >> > >> >> > >> >> Michael Jerris wrote: > >> >>> > >> >>> In short. No, you can not for many reasons. The milisecond tic is > >> >>> used throughout the code even when there is not any calls up. You > >> can > >> >>> grep for switch_cond_next if you would like to see where but it is > >> >>> required to keep our global timestamp and for pacing the scheduler > >> >>> among other services that run all the time. > >> >>> > >> >>> Mike > >> >>> > >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >>> > >> >>>> > >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >> >>>> glanced > >> >>>> through the code, and see that among others (are there others?) RTP > >> >>>> and IVR > >> >>>> set up their timers that are subsequently managed by this thread. > >> >>>> RTP timers > >> >>>> should be eliminated by that setting you've suggested. IVR timers > >> >>>> are set at > >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of > >> >>>> 1ms it > >> >>>> should be able to wake up those IVR timers just fine. Right? > >> >>>> > >> >>>> That's a cool design to have one dedicated thread that maintains > >> >>>> accurate > >> >>>> timing and then broadcasts via condition variables to hundreds of > >> >>>> other > >> >>>> threads events that they can register for. I'm sure it's one of the > >> >>>> reasons > >> >>>> why FS scales so much better than Asterisk. But for poor low-end > >> >>>> setups that > >> >>>> sit in the closet, eat only 6W of power and hardly ever run more > >> >>>> than two > >> >>>> calls at the same time, can I hack it somehow to be more UNIX- > >> >>>> friendly? I.e. > >> >>>> make it stuck in select() or recv() when there is nothing to do, > >> call > >> >>>> clock_gettime() right from the thread that wants and when it wants > >> >>>> to know > >> >>>> current time? > >> >>>> > >> >>>> Say, what if that thread is made to suspend on a condition variable > >> >>>> in case > >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some > >> other > >> >>>> thread comes up and adds its timer into the matrix, it could wake > up > >> >>>> the > >> >>>> timer thread and enjoy accurate timing as needed, on demand? And > in- > >> >>>> between > >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I > >> mean, > >> >>>> sitting on a wait queue in the kernel is way better than go back > and > >> >>>> forth > >> >>>> incrementing counters that nobody even needs at the moment? > >> >>>> > >> >>>> > >> >>>> Anthony Minessale-2 wrote: > >> >>>>> > >> >>>>> idle is a 4 letter word to a realtime application. > >> >>>>> > >> >>>>> The core keeps a single high-priority thread to keep 1ms timing > and > >> >>>>> expands > >> >>>>> that broadcasting > >> >>>>> to hundreds or thousand of threads who need accurate timing. > >> >>>>> > >> >>>>> Your choppy audio is caused by linksys lying about the packet len > >> >>>>> that > >> >>>>> it's > >> >>>>> using and we set our timer > >> >>>>> to the wrong speed. > >> >>>>> > >> >>>>> > >> >>>> > >> >>>> -- > >> >>>> View this message in context: > >> >>>> > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >>>> > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch- > >> >>>> users > >> >>>> http://www.freeswitch.org > >> >>> > >> >>> _______________________________________________ > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >>> > >> >> > >> >> > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/d9ff3e1b/attachment-0002.html From freeswitch-users-list at metik.com Thu Dec 3 10:17:43 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 03 Dec 2009 13:17:43 -0500 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: References: Message-ID: <4B1800C7.7010800@metik.com> Yehavi, There are a few variations of transmitting this information... If you have already enabled a supplemental isdn service profile, try adding the following to the PRI you are using: (config-if)#isdn outgoing ie facility (config-if)#iisdn outgoing ie extended-facility (config-if)#isdn outgoing display-ie (config-if)#isdn outgoing ie caller-number (config-if)#isdn outgoing ie called-number -metik Yehavi Bourvine wrote: > Hello, > > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On > the PRI there is a Nortel with Q.Sig. After a lot of configuration > trials I've managed to set it to send back the connected name over the > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the > connected name and then the Cisco adds it as a Remote-Party-ID). > However, I did not save it and a power outage cleared this config. In > my age I don't remember what I've done... > > Anyone knows the correct config? > > Thanks! __Yehavi: > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shiyanov at gmail.com Thu Dec 3 10:20:12 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Thu, 3 Dec 2009 21:20:12 +0300 Subject: [Freeswitch-users] uuid_bridge kills both channels if they are executing java app In-Reply-To: <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> References: <191c3a030911191849h3ba69116ob442d9712c2e74d2@mail.gmail.com> <191c3a030912021124n144075f6ne99a7c58c2e0f198@mail.gmail.com> Message-ID: I've sent deep-breath message to the dev list. Just-in-case, here is a "cross-post": Hi there! This message is a forward from user-mail-list. I'm trying to fix such a problem: FreSwithch compiled from SVN-trunk, date = 11/02/2009. What is need: connect two users, initially one is on the home-grown java-based IVR and other party is off hook. What is done/got: User1 is on the java application, it represents simple IVR system, and the most used FS API operation is "streamFile". User2 is off hook. next: (mod_socket) create_uuid bgapi originate {origination_caller_id_name=User1}[origination_uuid=uuid_x]User1 &park() uuid_bridge uuid_User1 uuid_User2 FS log is here: http://pastebin.freeswitch.org/11380 Thank you much for any help, Artem On Wed, Dec 2, 2009 at 10:24 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you should be working on SVN trunk if you are doing development, we are so > far forward from 1.0.4 we can't do debugging very easily. > > I don't know all of the details of what you are trying to do but you are > hitting some race conditions because of the async nature of the socket > connection and the way you are using it. > > > > > On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov wrote: > >> I'm back again with the same issue. >> Now it is became worse: it reproduces occasionally. >> [FS version is 1.04, test_load = 2 active calls] >> >> I've got 2 logs: successful and not. >> Here is a bad_case: >> >> 2009-12-02 13:27:55.159931 [NOTICE] switch_core_session.c:1576 Execute >> java(/usr/local/freeswitch/scripts/fs2agi.jar >> org.starpound.fs2agi.Translator >> ${agi_url}) >> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run >> INFO: *************************************************** >> Dec 2, 2009 1:27:55 PM org.starpound.fs2agi.Translator run >> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d >> 2009-12-02 13:27:55.169841 [NOTICE] switch_cpp.cpp:1130 Run AGI >> application >> agi://localhost:4573/hello.agi?callId=929 >> 2009-12-02 13:28:02.888831 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! >> >> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-12-02 13:28:04.799806 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/2001 [76d2c0e9-16a4-4098-92c0-5977cb482e17] >> 2009-12-02 13:28:05.148834 [NOTICE] sofia.c:3353 Ring-Ready >> sofia/internal/2001! >> 2009-12-02 13:28:05.855093 [NOTICE] sofia.c:3794 Channel >> [sofia/internal/2001] has >> been answered >> Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: >> java.lang.Exception: Internal FreeSwitch failure while streamming file, >> see >> FreeSwitch logs for details >> at >> >> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) >> at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) >> at org.starpound.fs2agi.Translator.run(Translator.java:56) >> at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) >> at >> >> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) >> at >> >> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) >> at java.lang.reflect.Method.invoke(Method.java:597) >> at org.freeswitch.Launcher.launch(Launcher.java:80) >> 2009-12-02 13:28:05.870185 [NOTICE] switch_core_state_machine.c:179 Hangup >> sofia/internal/2001 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-12-02 13:28:05.878807 [INFO] switch_cpp.cpp:1130 AGI application >> agi://localhost:4573/hello.agi?callId=929 crashed. See FS2AGI log for >> details. >> 2009-12-02 13:28:05.894422 [NOTICE] switch_ivr_bridge.c:667 Hangup >> sofia/external/6786081291 at 66.19.38.143 [CS_SOFT_EXECUTE] >> [DESTINATION_OUT_OF_ORDER] >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 17 >> (sofia/external/6786081291 at 66.19.38.143) Ended >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close >> Channel >> sofia/external/6786081291 at 66.19.38.143 [CS_DESTROY] >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1086 Session 18 >> (sofia/internal/2001) Ended >> 2009-12-02 13:28:05.894422 [NOTICE] switch_core_session.c:1088 Close >> Channel >> sofia/internal/2001 [CS_DESTROY] >> >> >> >> Message >> "Dec 2, 2009 1:28:05 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=929 for >> session >> 2898ad41-4ec1-4628-89fd-651a93a7221d crashed. Exception is: >> ..." >> is sent from my app upon the onHangup().` >> >> And here is a good_case: >> >> 2009-12-02 13:31:45.959813 [NOTICE] switch_core_session.c:1576 Execute >> java(/usr/local/freeswitch/scripts/fs2agi.jar >> org.starpound.fs2agi.Translator >> ${agi_url}) >> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run >> INFO: *************************************************** >> Dec 2, 2009 1:31:45 PM org.starpound.fs2agi.Translator run >> INFO: Run AGI application agi://localhost:4573/hello.agi?callId=932 for >> session >> 7c37369b-ffb2-4436-9288-a640047d0e5e >> 2009-12-02 13:31:45.965814 [NOTICE] switch_cpp.cpp:1130 Run AGI >> application >> agi://localhost:4573/hello.agi?callId=932 >> 2009-12-02 13:31:53.648915 [CRIT] mod_local_stream.c:234 Leaking stream >> handle! >> >> [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >> 2009-12-02 13:31:59.260797 [NOTICE] switch_channel.c:602 New Channel >> sofia/internal/2001 [7db554a6-861e-4492-a87b-78b6dfec6488] >> 2009-12-02 13:31:59.624818 [NOTICE] sofia.c:3353 Ring-Ready >> sofia/internal/2001! >> 2009-12-02 13:32:00.130814 [NOTICE] sofia.c:3794 Channel >> [sofia/internal/2001] has >> been answered >> Dec 2, 2009 1:32:00 PM org.starpound.fs2agi.Translator tellAllWeCrashed >> INFO: AGI application agi://localhost:4573/hello.agi?callId=932 for >> session >> 7c37369b-ffb2-4436-9288-a640047d0e5e crashed. Exception is: >> java.lang.Exception: Internal FreeSwitch failure while streamming file, >> see >> FreeSwitch logs for details >> at >> >> org.starpound.fs2agi.agicommands.StreamFileCommand.execute(StreamFileCommand.java:36) >> at org.starpound.fs2agi.AgiConnection.run(AgiConnection.java:48) >> at org.starpound.fs2agi.Translator.run(Translator.java:56) >> at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) >> at >> >> sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) >> at >> >> sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) >> at java.lang.reflect.Method.invoke(Method.java:597) >> at org.freeswitch.Launcher.launch(Launcher.java:80) >> 2009-12-02 13:32:00.149080 [INFO] switch_cpp.cpp:1130 AGI application >> agi://localhost:4573/hello.agi?callId=932 crashed. See FS2AGI log for >> details. >> 2009-12-02 13:32:00.388838 [INFO] switch_rtp.c:1869 Auto Changing port >> from >> 172.26.10.39:26402 to 91.190.120.190:26402 >> >> >> >> Suggestions? >> >> >> >> >> >> >> >> >> >> >> >> On Sat, Nov 21, 2009 at 12:58 PM, Artem Shiyanov wrote: >> >>> Anthony, >>> >>> >>As soon as you call uuid_bridge you are transferring both legs of the >>> call to bridge to each other. >>> >>This means your java app must exit so the channels can connect to each >>> other. >>> >>> I didn't know that. Now my java app is exiting upon the onHangup() call >>> so everything has become "ok". Thank you much. >>> I'll add note to the wiki about this issue. >>> >>> Artem >>> >>> >>> >>> >>> On Fri, Nov 20, 2009 at 5:49 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Your "annoying behaviour" is the exact behavior you should be getting >>>> considering what you told FS to do. >>>> >>>> As soon as you call uuid_bridge you are transferring both legs of the >>>> call to bridge to each other. >>>> This means your java app must exit so the channels can connect to each >>>> other. >>>> >>>> remember that you hangup hook can be called when the channel is >>>> transferred not only when it hangs up. >>>> you have to test which is happening based on the input to your callback. >>>> >>>> >>>> On Thu, Nov 19, 2009 at 1:46 PM, Artem Shiyanov wrote: >>>> >>>>> Hi there! >>>>> >>>>> I've got annoying FS behavior: >>>>> There are 2 channels executing the same Java application (application >>>>> itself is an IVR). If I try to bridge them with uuid_bridged then both >>>>> channels are killed. Here is a log from FS console: >>>>> uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_bridge.c:1165 >>>>> (sofia/internal/1005 at 192.168.147.130) State Change CS_EXECUTE -> >>>>> CS_HIBERNATE >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>>> called >>>>> 2009-07-09 05:58:26.562783 [DEBUG] switch_ivr_play_say.c:1391 done >>>>> playing file >>>>> 2009-07-09 05:58:26.576844 [DEBUG] switch_ivr_play_say.c:1391 done >>>>> playing file >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_core_session.c:933 Send >>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK] >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_ivr_bridge.c:1167 >>>>> (sofia/internal/1001 at master.agent.starpoundtech.net) State Change >>>>> CS_EXECUTE -> CS_HIBERNATE >>>>> 2009-07-09 05:58:26.641307 [DEBUG] switch_cpp.cpp:1185 hangup_hook >>>>> called >>>>> API CALL [uuid_bridge(68587a9d-1d20-48f1-bdfc-72a2c027e1d2 >>>>> 7d6c08fc-62bf-4a6c-a9ae-763d607e43de)] output: >>>>> +OK 7d6c08fc-62bf-4a6c-a9ae-763d607e43de >>>>> >>>>> freeswitch at localhost.localdomain> 2009-07-09 05:58:26.674348 [DEBUG] >>>>> switch_core_session.c:933 Send signal >>>>> sofia/internal/1001 at master.agent.starpoundtec >>>>> 2009-07-09 05:58:26.714809 [DEBUG] switch_core_session.c:813 Send >>>>> signal sofia/internal/1005 at 192.168.147.130 [BREAK] >>>>> >>>>> 2009-07-09 05:58:26.742764 [CRIT] mod_local_stream.c:234 Leaking stream >>>>> handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] >>>>> 2009-07-09 05:58:26.754791 [DEBUG] switch_core_session.c:813 Send >>>>> signal sofia/internal/1001 at master.agent.starpoundtech.net [BREAK] >>>>> >>>>> (FS version is 1.0.4) >>>>> >>>>> Any thoughts? >>>>> >>>>> >>>>> Artem >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/90a40da6/attachment-0002.html From dlaperle at rsslex.com Thu Dec 3 10:29:13 2009 From: dlaperle at rsslex.com (David Laperle) Date: Thu, 03 Dec 2009 13:29:13 -0500 Subject: [Freeswitch-users] Dialplan behavior Message-ID: <1259864953.1978.12.camel@dlaplap> Hi guys, i have a weird problem with my dialplans. For the moment, i have only 2 ??usable?? extensions. They were working #1 yesterday, but this morning i realize i forgot to compile mod_python, so i go back into my source folder and modify the modules.conf to uncomment mod_python, did a make and make install (i did a backup of my conf folder before)! The make and make install worked flawlessly. Then i put back my bkp of conf directory. I restarted the freeswitch service, created my python test dialplan and entered into cli to see what's gonna happen! To my surprise, the call didn't processed to the extension i was dialing. i tried all the other extensions i had, they were all not working!!!! After that i realized that the .xml in freeswitch/dialplan/default/ weren't imported into configuration at startup ... I have read all the documentation about difference between public and default dialplan and i understand them correctly, in public if i include all default folder, it's working again (i can reach all my extensions in default. My extensions are in the correct user_context ... i did nothing since yesterday other than a make && make install after enabling python ... Any other user have an idea why the default/*.xml aren't processed automatically? What could i have done wrong so they are no longer processed? Thanks a lot, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. Bien que ce courriel ait ?t? trait? contre les virus, il est de la responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos communications avec vous peuvent contenir des renseignements confidentiels ou prot?g?s par le secret professionnel. Si vous d?sirez que nous communiquions avec vous par un autre moyen de transmission que le courrier ?lectronique ordinaire non s?curis?, veuillez nous en aviser. NOTICE: This privileged and confidential email is intended only for the individual or entity to whom it is addressed. With regard to all others, any action related with this email as well as any reading, reproduction, transmission and/or dissemination in whole or in part of the information included in this email is prohibited. If you are not the addressee, immediately return the email to sender prior to destroying all copies. Even if this email is believed to be free from any virus, it is the responsibility of the recipient to make sure that it is virus exempt. Our communications to you may contain confidential information or information protected under solicitor-client privilege. Please advise if you wish us to use a mode of communication other than regular, unsecured e-mail in our communications with you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/1d1b11fa/attachment-0002.html From abeka at greatiam.com Thu Dec 3 10:34:25 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:34:25 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B1804B1.2060104@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From abeka at greatiam.com Thu Dec 3 10:34:45 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:34:45 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B1804C5.5070302@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From abeka at greatiam.com Thu Dec 3 10:36:21 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Thu, 03 Dec 2009 18:36:21 +0000 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> Message-ID: <4B180525.7060702@greatiam.com> Hi Sorry .xm is a typo. I actually shut down the server and restarted. The log says I need to create a domain of aaa.bbb.ccc.ddd (which is the server IP address ) and then put the user in that domain. Isn't the default domain that of the server FS is running on ? 2319.xml is in /usr/local/freeswitch/conf/directory/default/ Thanks for your time Michael Collins wrote: > > > On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah > > wrote: > > I have copied 1001.xml in directory/default to a test user 2319.xm > changing or instances of 1001 in the file to 2319. I then went into > default.xml in directory folder and in one of the groups just > mimicked > 1001 details by changing 1001 to 2319. > > Connecting to FS gives Forbidden message. However 1001 connects > without > a problem. What have I missed ? > > Is there a place that just puts things in do this and that and that to > create a new user ? > > > Did you execute "reloadxml" from the fs cli before trying to connect > with 2319? Also I'm assuming that "2319.xm" is a typo and you actually > created "2319.xml" in the default/directory subdir. > -MC > From erandr-junk at usa.net Thu Dec 3 10:43:40 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 10:43:40 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> Message-ID: <26630994.post@talk.nabble.com> You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I thought I responded back. Perhaps it didn't make through though, as I just emailed back to the list instead of using nabble.com... Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went w/o any effect either, but disabling RTP timers did the trick. I don't have the original "choppy sound with PCMU" problem any more, thanks a lot for the quick turnaround on that question. But your suggestions made me look, into logs, strace, code, etc, so now I'm just checking on how to quiet down those busy loops a little and how to get rid of periodic CRIT messages about Virtual Machine Migration. Anthony Minessale-2 wrote: > > What about the things I spent time suggesting in my last email? > Did you try them because I was actually curious if they made any impact. > > > On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > >> >> I'm sorry if I sounded that way. Did mean to. :) >> >> Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 >> chip >> and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm >> >> Line offset difference is due to some minor logging changes I made to see >> who's allocating timers and how often. This way I found MOH streaming and >> that RTP still allocates timers even when it's set to none in the >> profile. >> >> I feel that this platform turned out to be underpowered for FS because it >> cannot meet its scheduling expectations. I guess, some degree of kernel >> tweaking or setting priorities will fix that. Meanwhile I just got rid of >> the SQLDB 1ms thread via -nosql command line option, split sofia worker >> 1ms >> thread in two (one blocked and waiting for new commands in the SQL queue, >> the other one checking registrations and gateways with 1sec interval), >> and >> don't know yet what to do about the timer thread. >> >> Again, I apologize for stupid or accusing questions, I'm just trying to >> see >> how FS can be made friendlier to this board. Or the board be made >> friendlier >> to FS ;) >> >> >> Anthony Minessale-2 wrote: >> > >> > If you see that message then your machine/os/combo is having some >> problems >> > keeping up. >> > It's not the timer missing anything its the monotonic clock detecting a >> 1 >> > second or more differential from what its next prediction for the time >> > should be. The best way to trigger this would be to suspend FS with >> > control-z or attach to it with gdb blocking the entire process, that >> 1ms >> > thread would have to miss 1000 iterations to trigger that warning. >> > >> > Btw, that error message is at line 471 not 473 so you are using >> modified >> > code. >> > >> > Its possible your box has a bad monotonic timer, you can set >> > >> > >> > >> > under in switch.conf.xml >> > >> > We are now starting to guess you are using some small embedded type >> > platform >> > perhaps? >> > I've run FS even on a nokia n810 and never caused that message to fire. >> > >> > if 1 call can interrupt the cpu enough to cause noticeable issues you >> > might >> > want to consider running the process at a >> > greater priority by using the -hp command line arg or at least nice it >> > >> > Why don't you tell us the whole story about what OS/platform you are >> using >> > here rather that form conjectures about what is wrong with our code >> that >> > thousands of people are happy with. >> > >> > >> > >> > >> > >> > >> > >> > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: >> > >> >> >> >> Btw, I have these popping up in my logs from time to time: >> >> >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 >> >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP >> >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration >> >> Detected! Syncing Clock >> >> >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk >> >> picked >> >> up, but the surge of activity made FS timer thread miss a beat or two. >> >> >> >> >> >> eaf wrote: >> >> > >> >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps >> on >> >> > checking for messages every millisecond? Couldn't there be some >> >> signalling >> >> > implemented that will make the thread suspend on condition variable >> or >> >> a >> >> > socket/pipe in between? >> >> > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, >> obj=0x0) >> >> at >> >> > src/switch_core_sqldb.c:783 >> >> > >> >> > Why does this sofia_profile_worker_thread keeps on looping checking >> for >> >> > the queue? Have a semaphore! >> >> > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 >> >> > #1 0xb73a4701 in sofia_profile_worker_thread_run (thread=0x80f3a30, >> >> > obj=0x80f2490) at sofia.c:978 >> >> > >> >> > Nothing's happening on the box, but there are three threads that >> >> pretend >> >> > to be actively busy with smth. Others at least sleep for hundreds of >> >> > milliseconds, not for one. >> >> > >> >> > And there is even infrastructure present to do blocking pops: i.e. >> why >> >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() >> >> intermixed >> >> > with 1ms sleeps? This looping is such a waste... >> >> > >> >> > >> >> > eaf wrote: >> >> >> >> >> >> As I see it, switch_cond_next() currently is just a do_sleep(1000). >> >> Yes, >> >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" >> >> >> overrides that. >> >> >> >> >> >> Yeah, there is a global timestamp... It's easy to workaround that >> for >> >> RTP >> >> >> who calls switch_micro_time_now()... But if somebody accesses >> >> >> runtime.timestamp directly, it's gonna be tough to grep for that. >> If >> >> only >> >> >> this was C++... >> >> >> >> >> >> I'll play around. Never liked polling too much. Never could've >> guessed >> >> >> that polling could be so useful for scalability ;) My naive >> >> >> implementation would've pulled timestamp via system calls and >> would've >> >> >> done sleeping by passing exact interval to select() instead of >> syncing >> >> >> with a pacing thread. Which would be dead-quiet at idle time, but, >> of >> >> >> course, would stop scaling at some point due to excessive number of >> >> >> system calls. >> >> >> >> >> >> Thanks. >> >> >> >> >> >> >> >> >> Michael Jerris wrote: >> >> >>> >> >> >>> In short. No, you can not for many reasons. The milisecond tic is >> >> >>> used throughout the code even when there is not any calls up. You >> >> can >> >> >>> grep for switch_cond_next if you would like to see where but it is >> >> >>> required to keep our global timestamp and for pacing the scheduler >> >> >>> among other services that run all the time. >> >> >>> >> >> >>> Mike >> >> >>> >> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: >> >> >>> >> >> >>>> >> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I >> >> >>>> glanced >> >> >>>> through the code, and see that among others (are there others?) >> RTP >> >> >>>> and IVR >> >> >>>> set up their timers that are subsequently managed by this thread. >> >> >>>> RTP timers >> >> >>>> should be eliminated by that setting you've suggested. IVR timers >> >> >>>> are set at >> >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead of >> >> >>>> 1ms it >> >> >>>> should be able to wake up those IVR timers just fine. Right? >> >> >>>> >> >> >>>> That's a cool design to have one dedicated thread that maintains >> >> >>>> accurate >> >> >>>> timing and then broadcasts via condition variables to hundreds of >> >> >>>> other >> >> >>>> threads events that they can register for. I'm sure it's one of >> the >> >> >>>> reasons >> >> >>>> why FS scales so much better than Asterisk. But for poor low-end >> >> >>>> setups that >> >> >>>> sit in the closet, eat only 6W of power and hardly ever run more >> >> >>>> than two >> >> >>>> calls at the same time, can I hack it somehow to be more UNIX- >> >> >>>> friendly? I.e. >> >> >>>> make it stuck in select() or recv() when there is nothing to do, >> >> call >> >> >>>> clock_gettime() right from the thread that wants and when it >> wants >> >> >>>> to know >> >> >>>> current time? >> >> >>>> >> >> >>>> Say, what if that thread is made to suspend on a condition >> variable >> >> >>>> in case >> >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some >> >> other >> >> >>>> thread comes up and adds its timer into the matrix, it could wake >> up >> >> >>>> the >> >> >>>> timer thread and enjoy accurate timing as needed, on demand? And >> in- >> >> >>>> between >> >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I >> >> mean, >> >> >>>> sitting on a wait queue in the kernel is way better than go back >> and >> >> >>>> forth >> >> >>>> incrementing counters that nobody even needs at the moment? >> >> >>>> >> >> >>>> >> >> >>>> Anthony Minessale-2 wrote: >> >> >>>>> >> >> >>>>> idle is a 4 letter word to a realtime application. >> >> >>>>> >> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing >> and >> >> >>>>> expands >> >> >>>>> that broadcasting >> >> >>>>> to hundreds or thousand of threads who need accurate timing. >> >> >>>>> >> >> >>>>> Your choppy audio is caused by linksys lying about the packet >> len >> >> >>>>> that >> >> >>>>> it's >> >> >>>>> using and we set our timer >> >> >>>>> to the wrong speed. >> >> >>>>> >> >> >>>>> >> >> >>>> >> >> >>>> -- >> >> >>>> View this message in context: >> >> >>>> >> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html >> >> >>>> Sent from the Freeswitch-users mailing list archive at >> Nabble.com. >> >> >>>> >> >> >>>> >> >> >>>> _______________________________________________ >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch- >> >> >>>> users >> >> >>>> http://www.freeswitch.org >> >> >>> >> >> >>> _______________________________________________ >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >>> >> >> >> >> >> >> >> >> > >> >> > >> >> >> >> -- >> >> View this message in context: >> >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26630994.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mustafa.pk at gmail.com Thu Dec 3 10:46:44 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Thu, 3 Dec 2009 23:46:44 +0500 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <1259864953.1978.12.camel@dlaplap> References: <1259864953.1978.12.camel@dlaplap> Message-ID: <8213d6070912031046y6cd97f0cj38c9d392f92dfdda@mail.gmail.com> other than configuration/syntax problem it could be a simple character/file encoding problem or may be improper file permissions! On Thu, Dec 3, 2009 at 11:29 PM, David Laperle wrote: > Hi guys, > > i have a weird problem with my dialplans. For the moment, i have only 2 > ?usable? extensions. They were working #1 yesterday, but this morning i > realize i forgot to compile mod_python, so i go back into my source folder > and modify the modules.conf to uncomment mod_python, did a make and make > install (i did a backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test dialplan and > entered into cli to see what's gonna happen! To my surprise, the call didn't > processed to the extension i was dialing. > > i tried all the other extensions i had, they were all not working!!!! > > After that i realized that the .xml in freeswitch/dialplan/default/ weren't > imported into configuration at startup ... > > I have read all the documentation about difference between public and > default dialplan and i understand them correctly, in public if i include all > default folder, it's working again (i can reach all my extensions in > default. > > My extensions are in the correct user_context ... i did nothing since > yesterday other than a make && make install after enabling python ... > > Any other user have an idea why the default/*.xml aren't processed > automatically? What could i have done wrong so they are no longer processed? > > Thanks a lot, > > *David Laperle * > Administrateur r?seau / Network administrator > (514) 393-7647 > *dlaperle at rsslex.com* > > *Robinson Sheppard Shapiro *s.e.n.c.r.l/LLP > Avocats / Barristers & Solicitors > 4600 - 800 Place Victoria > Montr?al Qc H4Z 1H6 > T (514) 878-2631 F (514) 878-1865 > www.rsslex.com et/and www.rsscanadaimmigration.com > > > > > * > ------------------------------ > **http://www.rsslex.com** * > > *AVIS:* Ce courriel privil?gi? et confidentiel est destin? ? la seule > personne ou entit? ? laquelle il est adress?. Pour toute autre personne, > toute action prise en rapport ? ce courriel ainsi que toute lecture, > reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble > de celui-ci est interdite. Si vous n'?tes pas la personne autoris?e ? > recevoir ce courriel, S.V.P. le retourner ? l'exp?diteur et le d?truire. > Bien que ce courriel ait ?t? trait? contre les virus, il est de la > responsabilit? du destinataire de s'assurer que l'envoi en est exempt. Nos > communications avec vous peuvent contenir des renseignements confidentiels > ou prot?g?s par le secret professionnel. Si vous d?sirez que nous > communiquions avec vous par un autre moyen de transmission que le courrier > ?lectronique ordinaire non s?curis?, veuillez nous en aviser. > > *NOTICE:* This privileged and confidential email is intended only for the > individual or entity to whom it is addressed. With regard to all others, any > action related with this email as well as any reading, reproduction, > transmission and/or dissemination in whole or in part of the information > included in this email is prohibited. If you are not the addressee, > immediately return the email to sender prior to destroying all copies. Even > if this email is believed to be free from any virus, it is the > responsibility of the recipient to make sure that it is virus exempt. Our > communications to you may contain confidential information or information > protected under solicitor-client privilege. Please advise if you wish us to > use a mode of communication other than regular, unsecured e-mail in our > communications with you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/48c9e302/attachment-0002.html From asannucci at gmail.com Thu Dec 3 10:47:16 2009 From: asannucci at gmail.com (bakko) Date: Thu, 3 Dec 2009 13:47:16 -0500 Subject: [Freeswitch-users] can't register Inphonex In-Reply-To: <005a01ca73c9$bc2dcf60$34896e20$@ca> References: <005a01ca73c9$bc2dcf60$34896e20$@ca> Message-ID: <3A8174E906FB45CDA04B78C41ED21A88@voztovoice> >From de console: sofia profile external siptrace on or with ngrep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/ad12a5b6/attachment-0002.html From davis.erwin at gmail.com Thu Dec 3 11:30:49 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Thu, 3 Dec 2009 14:30:49 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: <191c3a030912021146v28d7b95es9a980f61613cdf8e@mail.gmail.com> <87f2f3b90912021303u1998aaf1rd4945a0dac5cc019@mail.gmail.com> Message-ID: Hi, Anthony and Mike, With the latest version from SVN, I was able to remove the warning "sample rate not matching". But the remote RTP port was still changed after after playing the vm greeting. See below, 2009-12-03 13:44:46.901216 [INFO] switch_rtp.c:1975 Auto Changing port from XXX.YYY.ZZZ.39:10002 to XXX.YYY.ZZZ.39:3335 Any clue? I looked at the source code in switch_rtp.c:1975, it shows that if rtp_session->autoadj_tally >= 10, then a rtp port change will happen. Any idea about autoadj_tally and what cause the increase of autoadj_tally ? Thanks, On 12/2/09, Erwin Davis wrote: > > Hi, Anthony and Mike, > > Thanks for your reply. The problem still exists even after I ran "make > hd-sounds install". > I will try the latest version from the SVN to see if the problem will go > away. I will let you know. > Thanks folks, > > Regards, > > On 12/2/09, Michael Collins wrote: > >> >> >> On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis wrote: >> >>> Hi, Anthony, >>> >>> Thanks for your reply. >>> >>> When I type the command below, I got the error, >>> Unknown target hd-sound-install >>> make[1]: *** [hd-sound-install] Error 1 >>> make: *** [hd-sound-install] Error 2 >>> >>> I found out that under >>> /usr/local/freeswitch/sounds/en/us/callie/voicemail, there are directories, >>> 8000, 16000, 32000, 48000 for recorded voicemail greetings. It should >>> explain why at first FS played in right sample rate. But after playing >>> serveral time, FS complained about sample rate not matching. Any clue? >>> Thanks, >>> >>> >> Erwin, >> >> As Tony said you've actually got a pretty old installation. If this is in >> production then I would recommend getting a sandbox machine, install trunk >> using the quick-and-dirty install, and then update the default config to you >> specific configuration. Test to make sure it works before you put it into >> production. :) >> >> Feel free to join us on IRC (#freeswitch on irc.freenode.net) if you run >> into any issues that require more real-time conversation. >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/447f2cfa/attachment-0002.html From yehavi.bourvine at gmail.com Thu Dec 3 11:31:57 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 3 Dec 2009 21:31:57 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: <4B1800C7.7010800@metik.com> References: <4B1800C7.7010800@metik.com> Message-ID: Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back to ISDN the name of the destination, but not the other way around... Thanks! __Yehavi: 2009/12/3 Metik > Yehavi, > > There are a few variations of transmitting this information... If you > have already enabled a supplemental isdn service profile, try adding the > following to the PRI you are using: > > (config-if)#isdn outgoing ie facility > (config-if)#iisdn outgoing ie extended-facility > (config-if)#isdn outgoing display-ie > (config-if)#isdn outgoing ie caller-number > (config-if)#isdn outgoing ie called-number > > -metik > > Yehavi Bourvine wrote: > > Hello, > > > > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On > > the PRI there is a Nortel with Q.Sig. After a lot of configuration > > trials I've managed to set it to send back the connected name over the > > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the > > connected name and then the Cisco adds it as a Remote-Party-ID). > > However, I did not save it and a power outage cleared this config. In > > my age I don't remember what I've done... > > > > Anyone knows the correct config? > > > > Thanks! __Yehavi: > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/998e13cf/attachment-0002.html From msc at freeswitch.org Thu Dec 3 11:51:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 11:51:09 -0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B1804B1.2060104@greatiam.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> <4B1804B1.2060104@greatiam.com> Message-ID: <87f2f3b90912031151kf2a6843w1b48cf36330a9252@mail.gmail.com> On Thu, Dec 3, 2009 at 10:34 AM, Samuel Abekah-Mensah wrote: > Hi > > Sorry .xm is a typo. I actually shut down the server and restarted. The > log says I need to create a domain of aaa.bbb.ccc.ddd (which is the > server IP address ) and then put the user in that domain. Isn't the > default domain that of the server FS is running on ? > 2319.xml is in /usr/local/freeswitch/conf/directory/default/ > > Thanks for your time > > Okay, here's exactly what I did: cd /usr/local/freeswitch/conf/directory/default cp 1001.xml 2319.xml perl -pi -e 's/1001/2319/g' 2319.xml cat 2319.xml Then I logged into fs_cli, pressed F6 (which does "reloadxml") and then I set up my x-lite: Display Name: Test User name: 2319 Password: 1234 Authorization user name: 2319 Domain: 10.15.0.91 It registered just fine as can be seen by the output of "sofia status profile internal": Call-ID: MzRiOGI4NTA2YjA0ZTkzMDYwZjA3MTlkZGQ3ZjNhMjg. User: 2319 at 10.15.0.91 Contact: "Test" Agent: X-Lite release 1014k stamp 47051 Status: Registered(UDP)(unknown) EXP(2009-12-03 13:41:38) Host: freeswitch1.yt IP: 10.15.0.124 Port: 41680 Auth-User: 2319 Auth-Realm: 10.15.0.91 MWI-Account: 2319 at 10.15.0.91 So, most likely you've got an issue with the XML file itself or the configuration on your SIP device. Double check the username and auth username values. If need be delete your 2319.xml file and start over. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/cb357bb8/attachment-0002.html From msc at freeswitch.org Thu Dec 3 11:55:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Dec 2009 11:55:23 -0800 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <1259864953.1978.12.camel@dlaplap> References: <1259864953.1978.12.camel@dlaplap> Message-ID: <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> On Thu, Dec 3, 2009 at 10:29 AM, David Laperle wrote: > Hi guys, > > i have a weird problem with my dialplans. For the moment, i have only 2 > ?usable? extensions. They were working #1 yesterday, but this morning i > realize i forgot to compile mod_python, so i go back into my source folder > and modify the modules.conf to uncomment mod_python, did a make and make > install (i did a backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test dialplan and > entered into cli to see what's gonna happen! To my surprise, the call didn't > processed to the extension i was dialing. > > i tried all the other extensions i had, they were all not working!!!! > > After that i realized that the .xml in freeswitch/dialplan/default/ weren't > imported into configuration at startup ... > > I have read all the documentation about difference between public and > default dialplan and i understand them correctly, in public if i include all > default folder, it's working again (i can reach all my extensions in > default. > > My extensions are in the correct user_context ... i did nothing since > yesterday other than a make && make install after enabling python ... > > Any other user have an idea why the default/*.xml aren't processed > automatically? What could i have done wrong so they are no longer processed? > > double-check for the existence of conf/dialplan/default.xml - I've seen on rare occasion where that file simple goes away for no apparent reason. Since I never change that file - and I recommend that you never change it either ;) - you can go to your FS source directory and issue "make samples" and it will re-create any missing default config files without overwriting you existing config files. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/6a7af16f/attachment-0002.html From jbr at consiglia.dk Thu Dec 3 12:05:17 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Thu, 3 Dec 2009 21:05:17 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I am trying to rewrite all my javascript scripts into Lua scripts. I have run into the problem of core_db access. This can be achieved with Spidermonkey, but apparently not with Lua. I have tried to get the binary for Lua (using apt-get) but I get an error when I require the sqlite.so: undefined symbol: luaopen_luasql_sqlite, so I'm stuck. So what is a feasible way to manipulate the core database from Lua? I may mention that access to MySQL works perfectly from Lua. Regards Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/43f1b4ac/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 12:29:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 14:29:13 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912031229g7b6bf3cdo43b58c43cd2be6a8@mail.gmail.com> In latest trunk you can run the core db in your same mysql db. other than that we would need to create an object from our lua module similar to how it was done in js. On Thu, Dec 3, 2009 at 2:05 PM, Jon Bruel wrote: > I am trying to rewrite all my javascript scripts into Lua scripts. I have > run into the problem of core_db access. This can be achieved with > Spidermonkey, but apparently not with Lua. I have tried to get the binary > for Lua (using apt-get) but I get an error when I require the sqlite.so: > undefined symbol: luaopen_luasql_sqlite, so I?m stuck. So what is a feasible > way to manipulate the core database from Lua? > > I may mention that access to MySQL works perfectly from Lua. > > Regards Jon > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/5d1a37cd/attachment-0002.html From timuckun at gmail.com Thu Dec 3 12:40:16 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 09:40:16 +1300 Subject: [Freeswitch-users] HA questions. In-Reply-To: <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> Message-ID: <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: > The easiest place to do this is at the point you send the calls to FreeSWITCH. ?How are the calls coming in? > >From an as of now unkown SIP trunk provider (we are still in negotiations with a couple of companies). From davis.erwin at gmail.com Thu Dec 3 12:41:00 2009 From: davis.erwin at gmail.com (Erwin Davis) Date: Thu, 3 Dec 2009 15:41:00 -0500 Subject: [Freeswitch-users] change the remote RTP port after sample rate doesnot match In-Reply-To: References: Message-ID: Hi, I solved this issue. the reason is because of the different port number between the the one in SDP and the one in real RTP stream. This is very nice feature. e On 12/2/09, Erwin Davis wrote: > > Hi, I got a weird issue when I dialed an extension and listen to a recorded > voice mail greeting message. > After playing a couple of time of the greeting, the FS printed the warning > of "sample rate not matching", then > send the audio to a different remote RTP port. See the log below, > > > 2009-12-02 12:42:27.109529 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:27.112038 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:28.86119 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:28.205975 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:28.220967 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-record_message.wav] (en:en) > 2009-12-02 12:42:28.222971 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:28.222971 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:32.804858 [DEBUG] switch_ivr_play_say.c:1391 done playing > file > 2009-12-02 12:42:32.944554 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [TRANSCODING_NECESSARY] > 2009-12-02 12:42:33.946969 [WARNING] switch_core_file.c:120 Sample rate > doesn't match > 2009-12-02 12:42:33.950231 [DEBUG] switch_ivr_play_say.c:549 Raw Codec > Activated > 2009-12-02 12:42:34.124584 [INFO] switch_rtp.c:1869 Auto Changing port from > xxx.yyy.zzz.39:10002 to xxx.yyy.zzz.39:1748 > 2009-12-02 12:42:34.944913 [DEBUG] switch_core_codec.c:122 Restore original > codec. > 2009-12-02 12:42:34.947918 [DEBUG] mod_voicemail.c:1162 Message is less > than minimum record length: 3, discarding it. > 2009-12-02 12:42:34.947918 [DEBUG] switch_ivr_play_say.c:118 No language > specified - Using [en] > 2009-12-02 12:42:34.960002 [DEBUG] switch_ivr_play_say.c:273 Handle > play-file:[voicemail/vm-too-small.wav] (en:en) > 2009-12-02 12:42:34.960850 [DEBUG] switch_ivr_play_say.c:1097 Codec > Activated L16 at 16000hz 1 channels 20ms > 2009-12-02 12:42:34.960850 [DEBUG] switch_core_io.c:649 > sofia/internal/1003 at xxx.yyy.zzz.31 receive message [ > > > the original codec is wideband 16kHz Speex and the wireshark shows that the > FS used the same codec. I used FS 1.04 in fedora 8. > I have two questions here, > (1) why does FS report "Sample rate doesn't match"? is it a bug or > configuration issue? > (2) Why does FS change the RTP port ? how to fix it? > > Thanks, > > Regards, > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/8b5bd90c/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 3 12:40:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 14:40:59 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26630994.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> Message-ID: <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> no, I mean the one after that that you must have completely skipped with a command line option to try and a param to set in the config. It somewhat annoys me for taking the time to compose it now. I wrote all of the code you are talking about myself and I was trying to give you some suggestions.... Well, actually, you did answer my question about the platform so you must have seen it..... The loops are not the cause of that migration message, something wrong with the hardware or the kernel is. Another guy just told you he does not see that problem on the same exact hardware. Even if you have a point about the sql threads, you could make a patch to slow them down but you cant slow down too much or you will not be able to handle 400 cps all asking to send updates to transactions in batches of thousands of sql stmts. Every line of that code is carefully designed so I don't know what else to tell you but to stop being so arrogant and re-read this thread for all the advice you have totally ignored. I started out trying to help you but I have a lot of work to do. I thoroughly explained it to you and you are choosing to ignore me so I guess I'm done. You can do whatever you want with your working copy, i'll see you in 3 or 4 years when you get up to speed with the rest of us........ On Thu, Dec 3, 2009 at 12:43 PM, eaf wrote: > > You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I > thought I responded back. Perhaps it didn't make through though, as I just > emailed back to the list instead of using nabble.com... > > Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went > w/o any effect either, but disabling RTP timers did the trick. I don't have > the original "choppy sound with PCMU" problem any more, thanks a lot for > the > quick turnaround on that question. > > But your suggestions made me look, into logs, strace, code, etc, so now I'm > just checking on how to quiet down those busy loops a little and how to get > rid of periodic CRIT messages about Virtual Machine Migration. > > > Anthony Minessale-2 wrote: > > > > What about the things I spent time suggesting in my last email? > > Did you try them because I was actually curious if they made any impact. > > > > > > On Thu, Dec 3, 2009 at 11:29 AM, eaf wrote: > > > >> > >> I'm sorry if I sounded that way. Did mean to. :) > >> > >> Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 > >> chip > >> and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm > >> > >> Line offset difference is due to some minor logging changes I made to > see > >> who's allocating timers and how often. This way I found MOH streaming > and > >> that RTP still allocates timers even when it's set to none in the > >> profile. > >> > >> I feel that this platform turned out to be underpowered for FS because > it > >> cannot meet its scheduling expectations. I guess, some degree of kernel > >> tweaking or setting priorities will fix that. Meanwhile I just got rid > of > >> the SQLDB 1ms thread via -nosql command line option, split sofia worker > >> 1ms > >> thread in two (one blocked and waiting for new commands in the SQL > queue, > >> the other one checking registrations and gateways with 1sec interval), > >> and > >> don't know yet what to do about the timer thread. > >> > >> Again, I apologize for stupid or accusing questions, I'm just trying to > >> see > >> how FS can be made friendlier to this board. Or the board be made > >> friendlier > >> to FS ;) > >> > >> > >> Anthony Minessale-2 wrote: > >> > > >> > If you see that message then your machine/os/combo is having some > >> problems > >> > keeping up. > >> > It's not the timer missing anything its the monotonic clock detecting > a > >> 1 > >> > second or more differential from what its next prediction for the time > >> > should be. The best way to trigger this would be to suspend FS with > >> > control-z or attach to it with gdb blocking the entire process, that > >> 1ms > >> > thread would have to miss 1000 iterations to trigger that warning. > >> > > >> > Btw, that error message is at line 471 not 473 so you are using > >> modified > >> > code. > >> > > >> > Its possible your box has a bad monotonic timer, you can set > >> > > >> > > >> > > >> > under in switch.conf.xml > >> > > >> > We are now starting to guess you are using some small embedded type > >> > platform > >> > perhaps? > >> > I've run FS even on a nokia n810 and never caused that message to > fire. > >> > > >> > if 1 call can interrupt the cpu enough to cause noticeable issues you > >> > might > >> > want to consider running the process at a > >> > greater priority by using the -hp command line arg or at least nice it > >> > > >> > Why don't you tell us the whole story about what OS/platform you are > >> using > >> > here rather that form conjectures about what is wrong with our code > >> that > >> > thousands of people are happy with. > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > On Thu, Dec 3, 2009 at 8:55 AM, eaf wrote: > >> > > >> >> > >> >> Btw, I have these popping up in my logs from time to time: > >> >> > >> >> 2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314 > >> >> (sofia/external/xxxxx at 4.68.250.148) Running State Change CS_HANGUP > >> >> 2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration > >> >> Detected! Syncing Clock > >> >> > >> >> In this case an incoming call rang to both FS and Asterisk, Asterisk > >> >> picked > >> >> up, but the surge of activity made FS timer thread miss a beat or > two. > >> >> > >> >> > >> >> eaf wrote: > >> >> > > >> >> > Oh, it's not just one timer thread... Why, why is sql_thread keeps > >> on > >> >> > checking for messages every millisecond? Couldn't there be some > >> >> signalling > >> >> > implemented that will make the thread suspend on condition variable > >> or > >> >> a > >> >> > socket/pipe in between? > >> >> > > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> >> > #1 0xb7e5e215 in switch_core_sql_thread (thread=0xb7586ae8, > >> obj=0x0) > >> >> at > >> >> > src/switch_core_sqldb.c:783 > >> >> > > >> >> > Why does this sofia_profile_worker_thread keeps on looping checking > >> for > >> >> > the queue? Have a semaphore! > >> >> > > >> >> > #0 do_sleep (t=1000) at src/switch_time.c:109 > >> >> > #1 0xb73a4701 in sofia_profile_worker_thread_run > (thread=0x80f3a30, > >> >> > obj=0x80f2490) at sofia.c:978 > >> >> > > >> >> > Nothing's happening on the box, but there are three threads that > >> >> pretend > >> >> > to be actively busy with smth. Others at least sleep for hundreds > of > >> >> > milliseconds, not for one. > >> >> > > >> >> > And there is even infrastructure present to do blocking pops: i.e. > >> why > >> >> > couldn't sqldb thread do queue_pop() instead of queue_trypop() > >> >> intermixed > >> >> > with 1ms sleeps? This looping is such a waste... > >> >> > > >> >> > > >> >> > eaf wrote: > >> >> >> > >> >> >> As I see it, switch_cond_next() currently is just a > do_sleep(1000). > >> >> Yes, > >> >> >> it could be mapped to a 1ms timer, but "#define DISABLE_1MS_COND" > >> >> >> overrides that. > >> >> >> > >> >> >> Yeah, there is a global timestamp... It's easy to workaround that > >> for > >> >> RTP > >> >> >> who calls switch_micro_time_now()... But if somebody accesses > >> >> >> runtime.timestamp directly, it's gonna be tough to grep for that. > >> If > >> >> only > >> >> >> this was C++... > >> >> >> > >> >> >> I'll play around. Never liked polling too much. Never could've > >> guessed > >> >> >> that polling could be so useful for scalability ;) My naive > >> >> >> implementation would've pulled timestamp via system calls and > >> would've > >> >> >> done sleeping by passing exact interval to select() instead of > >> syncing > >> >> >> with a pacing thread. Which would be dead-quiet at idle time, but, > >> of > >> >> >> course, would stop scaling at some point due to excessive number > of > >> >> >> system calls. > >> >> >> > >> >> >> Thanks. > >> >> >> > >> >> >> > >> >> >> Michael Jerris wrote: > >> >> >>> > >> >> >>> In short. No, you can not for many reasons. The milisecond tic > is > >> >> >>> used throughout the code even when there is not any calls up. > You > >> >> can > >> >> >>> grep for switch_cond_next if you would like to see where but it > is > >> >> >>> required to keep our global timestamp and for pacing the > scheduler > >> >> >>> among other services that run all the time. > >> >> >>> > >> >> >>> Mike > >> >> >>> > >> >> >>> On Dec 2, 2009, at 7:31 PM, eaf wrote: > >> >> >>> > >> >> >>>> > >> >> >>>> Can I reduce resolution of that timer thread 10 times? I mean, I > >> >> >>>> glanced > >> >> >>>> through the code, and see that among others (are there others?) > >> RTP > >> >> >>>> and IVR > >> >> >>>> set up their timers that are subsequently managed by this > thread. > >> >> >>>> RTP timers > >> >> >>>> should be eliminated by that setting you've suggested. IVR > timers > >> >> >>>> are set at > >> >> >>>> 20ms... So, if the thread is set to wake up every 10ms instead > of > >> >> >>>> 1ms it > >> >> >>>> should be able to wake up those IVR timers just fine. Right? > >> >> >>>> > >> >> >>>> That's a cool design to have one dedicated thread that maintains > >> >> >>>> accurate > >> >> >>>> timing and then broadcasts via condition variables to hundreds > of > >> >> >>>> other > >> >> >>>> threads events that they can register for. I'm sure it's one of > >> the > >> >> >>>> reasons > >> >> >>>> why FS scales so much better than Asterisk. But for poor low-end > >> >> >>>> setups that > >> >> >>>> sit in the closet, eat only 6W of power and hardly ever run more > >> >> >>>> than two > >> >> >>>> calls at the same time, can I hack it somehow to be more UNIX- > >> >> >>>> friendly? I.e. > >> >> >>>> make it stuck in select() or recv() when there is nothing to do, > >> >> call > >> >> >>>> clock_gettime() right from the thread that wants and when it > >> wants > >> >> >>>> to know > >> >> >>>> current time? > >> >> >>>> > >> >> >>>> Say, what if that thread is made to suspend on a condition > >> variable > >> >> >>>> in case > >> >> >>>> if there are no timers registered in TIMER_MATRIX? Then, if some > >> >> other > >> >> >>>> thread comes up and adds its timer into the matrix, it could > wake > >> up > >> >> >>>> the > >> >> >>>> timer thread and enjoy accurate timing as needed, on demand? And > >> in- > >> >> >>>> between > >> >> >>>> the calls, when there is no RTP or IVR, it will all go silent? I > >> >> mean, > >> >> >>>> sitting on a wait queue in the kernel is way better than go back > >> and > >> >> >>>> forth > >> >> >>>> incrementing counters that nobody even needs at the moment? > >> >> >>>> > >> >> >>>> > >> >> >>>> Anthony Minessale-2 wrote: > >> >> >>>>> > >> >> >>>>> idle is a 4 letter word to a realtime application. > >> >> >>>>> > >> >> >>>>> The core keeps a single high-priority thread to keep 1ms timing > >> and > >> >> >>>>> expands > >> >> >>>>> that broadcasting > >> >> >>>>> to hundreds or thousand of threads who need accurate timing. > >> >> >>>>> > >> >> >>>>> Your choppy audio is caused by linksys lying about the packet > >> len > >> >> >>>>> that > >> >> >>>>> it's > >> >> >>>>> using and we set our timer > >> >> >>>>> to the wrong speed. > >> >> >>>>> > >> >> >>>>> > >> >> >>>> > >> >> >>>> -- > >> >> >>>> View this message in context: > >> >> >>>> > >> >> > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26619085.html > >> >> >>>> Sent from the Freeswitch-users mailing list archive at > >> Nabble.com. > >> >> >>>> > >> >> >>>> > >> >> >>>> _______________________________________________ > >> >> >>>> FreeSWITCH-users mailing list > >> >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >>>> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch- > >> >> >>>> users > >> >> >>>> http://www.freeswitch.org > >> >> >>> > >> >> >>> _______________________________________________ > >> >> >>> FreeSWITCH-users mailing list > >> >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >>> UNSUBSCRIBE: > >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org > >> >> >>> > >> >> >>> > >> >> >> > >> >> >> > >> >> > > >> >> > > >> >> > >> >> -- > >> >> View this message in context: > >> >> > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26627246.html > >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >> > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > -- > >> > Anthony Minessale II > >> > > >> > FreeSWITCH http://www.freeswitch.org/ > >> > ClueCon http://www.cluecon.com/ > >> > Twitter: http://twitter.com/FreeSWITCH_wire > >> > > >> > AIM: anthm > >> > MSN:anthony_minessale at hotmail.com > >> > >< > >> MSN%3Aanthony_minessale at hotmail.com > > > > >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > > > > >> > > >> > IRC: irc.freenode.net #freeswitch > >> > > >> > FreeSWITCH Developer Conference > >> > sip:888 at conference.freeswitch.org > >> > >< > >> sip%3A888 at conference.freeswitch.org > > > > >> > > >> > iax:guest at conference.freeswitch.org/888 > >> > > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> > > > > >> > > >> > pstn:213-799-1400 > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26629856.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26630994.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/b872350e/attachment-0002.html From timuckun at gmail.com Thu Dec 3 12:41:49 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 09:41:49 +1300 Subject: [Freeswitch-users] HA questions. In-Reply-To: <012801ca7439$8cc10ba0$a64322e0$@net> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <012801ca7439$8cc10ba0$a64322e0$@net> Message-ID: <855e4dcf0912031241l73ea409fx71f8e0b7b0b79239@mail.gmail.com> On Fri, Dec 4, 2009 at 5:56 AM, Adam Ford wrote: > Have you checked out Redfone? While I haven't attempted to implement it yet, > my Redfone foneBridge2 claims to be able to handle load balancing and > failover between two Asterisk/Freeswitch servers. > That would be my choice for incoming E1 lines. Right now I am looking for a SIP solution. From dlaperle at rsslex.com Thu Dec 3 13:04:03 2009 From: dlaperle at rsslex.com (David Laperle) Date: Thu, 03 Dec 2009 16:04:03 -0500 Subject: [Freeswitch-users] Dialplan behavior In-Reply-To: <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> References: <1259864953.1978.12.camel@dlaplap> <87f2f3b90912031155mb69783tdc298f01f57f8cb3@mail.gmail.com> Message-ID: <1259874243.8702.9.camel@dlaplap> The files are OK, the permissions on them are OK. Correct me if i'm wrong! If i set the variable "user_context" to "default" it should take into accounts the dialplans into freeswitch/conf/dialplan/default or there's more rules to consider? The Wiki explain that the user must be registered to receive the xml in diaplan/default, my phone is registered (see below) Call-ID: 7cd6e8c8-2c9962a5-7a0da12a at 192.168.102.10 User: 6969@ Contact: "user" Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.2.0078 Status: Registered(UDP)(unknown) EXP(2009-12-03 16:57:35) Host: IP: 192.168.102.10 Port: 5060 Auth-User: unknown Auth-Realm: MWI-Account: 6969@ The only thing i can think of is that my installation of ??FusionPBX?? (installed since day 1) messed something in config files but i really doubt it since i didn't use FusionPBX between yesterday (when was working good) and this morning (after python recompile). The other thing i see is the ??Auth-User?? in my registrations who shows ??unknown?? instead of the actual user ... I'm really lost since everything was working perfectly before my re-compile of mod_python, i think i'll just start all over again ... since i had almost nothing done so far! Thanks for your time, David Laperle Administrateur r??seau / Network administrator (514) 393-7647 dlaperle at rsslex.com Robinson Sheppard Shapiro s.e.n.c.r.l/LLP Avocats / Barristers & Solicitors 4600 - 800 Place Victoria Montr??al Qc H4Z 1H6 T (514) 878-2631 F (514) 878-1865 www.rsslex.com et/and www.rsscanadaimmigration.com On Thu, 2009-12-03 at 14:55 -0500, Michael Collins wrote: > > > > On Thu, Dec 3, 2009 at 10:29 AM, David Laperle > wrote: > > Hi guys, > > i have a weird problem with my dialplans. For the moment, i > have only 2 usable extensions. They were working #1 yesterday, > but this morning i realize i forgot to compile mod_python, so > i go back into my source folder and modify the modules.conf to > uncomment mod_python, did a make and make install (i did a > backup of my conf folder before)! The make and make install > worked flawlessly. Then i put back my bkp of conf directory. > > I restarted the freeswitch service, created my python test > dialplan and entered into cli to see what's gonna happen! To > my surprise, the call didn't processed to the extension i was > dialing. > > i tried all the other extensions i had, they were all not > working!!!! > > After that i realized that the .xml in > freeswitch/dialplan/default/ weren't imported into > configuration at startup ... > > I have read all the documentation about difference between > public and default dialplan and i understand them correctly, > in public if i include all default folder, it's working again > (i can reach all my extensions in default. > > My extensions are in the correct user_context ... i did > nothing since yesterday other than a make && make install > after enabling python ... > > Any other user have an idea why the default/*.xml aren't > processed automatically? What could i have done wrong so they > are no longer processed? > > > > double-check for the existence of conf/dialplan/default.xml - I've > seen on rare occasion where that file simple goes away for no apparent > reason. Since I never change that file - and I recommend that you > never change it either ;) - you can go to your FS source directory and > issue "make samples" and it will re-create any missing default config > files without overwriting you existing config files. > -MC > > > -------------------------------------------------------------------------------- http://www.rsslex.com AVIS: Ce courriel privil?gi? et confidentiel est destin? ? la seule personne ou entit? ? laquelle il est adress?. Pour toute autre personne, toute action prise en rapport ? ce courriel ainsi que toute lecture, reproduction, transmission et/ou divulgation d'une partie ou de l'ensemble de celui-ci est interdite. 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/c590d9cc/attachment-0002.html From erandr-junk at usa.net Thu Dec 3 13:44:15 2009 From: erandr-junk at usa.net (eaf) Date: Thu, 3 Dec 2009 13:44:15 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> Message-ID: <26633739.post@talk.nabble.com> Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do that. At the moment, I hope it won't be necessary as I can make those "hyper" threads behave, and will see how that goes first. I see where your implementation could be coming from. There is a queue of SQL queries in sofia.c processed by the worker thread. There are only two pop functions available in APR: queue_pop() and queue_trypop(), so alas no option with a timeout here. You don't want to block the thread in pop() indefinitely because you chose that same worker needs to do ireg and gw processing once in a while (separated by tens or hundreds of seconds, btw). You also want to be able to detect shutdown condition so that the worker doesn't hold up profile thread. So you chose to poll for events every millisecond instead of just creating an apr_thread_cond_t for resource friendly signalling. I agree that the timer thread philosophy is great and was the right choice for scaling, but I just don't comprehend responses to things like these other SQL or sofia worker threads. Did somebody even remotely acknowledge that busy loops at least in those areas that I showed may probably be a bad idea and could've been eliminated? I've heard suggestions to bump up priority, I've heard that the code was perfect already, that it's the result of 4-year effort, that I am arrogant, don't listen and don't understand squat. I'm sorry if I gave you impression that I was looking for the bad parts in the software. I apologized for that already. All I wanted was to have constructive conversation, perhaps I'm not too good at it. Code is already perfect according to you? Fine with me. Anthony Minessale-2 wrote: > > no, > > I mean the one after that that you must have completely skipped with a > command line option to try and a param to set in the config. It somewhat > annoys me for taking the time to compose it now. I wrote all of the code > you are talking about myself and I was trying to give you some > suggestions.... > > Well, actually, you did answer my question about the platform so you must > have seen it..... > > The loops are not the cause of that migration message, something wrong > with > the hardware or the kernel is. > Another guy just told you he does not see that problem on the same exact > hardware. > > Even if you have a point about the sql threads, you could make a patch to > slow them down but you cant slow down too much or you will not be able to > handle 400 cps all asking to send updates to transactions in batches of > thousands of sql stmts. Every line of that code is carefully designed so > I > don't know what else to tell you but to stop being so arrogant and re-read > this thread for all the advice you have totally ignored. I started out > trying to help you but I have a lot of work to do. I thoroughly explained > it to you and you are choosing to ignore me so I guess I'm done. > You can do whatever you want with your working copy, i'll see you in 3 or > 4 > years when you get up to speed with the rest of us........ > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mcampbellsmith at gmail.com Thu Dec 3 14:05:35 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 09:05:35 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> Message-ID: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Hi All, I managed to borrow a SPA3102 with the latest firmware and have got it to register using TLS, but I am still struggling with SRTP. Has anyone managed to get SRTP working with the Linksys devices and if so, can they direct me on how to do this. I have generated a mini-certificates and SRTP Private Key using the gen-mc tool found at http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. However, when ever I initiate a call from the SPA, I can see that the call is not encrypted. Help appreciated. Thanks! On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > Check out the Linksys SPA2102 > > On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > wrote: >> >> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> Grandstream HandyTone 503. ?But, again according to the wiki, that >> doesn't seem to behave to well with TLS ... >> >> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >> > Mark Campbell-Smith wrote: >> >> Does the SPA3102 support TLS or only SRTP? >> > >> > I don't know, but supporting only SRTP would be ridiculous, since the >> > keys >> > would then be transmitted in the clear and therefore amenable to >> > interception. >> > SRTP requires the SIP channel to be encrypted by TLS in order to be >> > secure. >> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >> > in >> > RTP. >> > >> > I would be rather surprised were a hardware manufacturer to implement >> > SRTP >> > without TLS for the SIP traffic. On the other hand, we've seen often in >> > this >> > forum that some manufacturers are really clueless... >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From itamar at ispbrasil.com.br Thu Dec 3 14:17:14 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Thu, 3 Dec 2009 20:17:14 -0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Message-ID: you can try xlite too. On Thu, Dec 3, 2009 at 8:05 PM, Mark Campbell-Smith wrote: > Hi All, > > I managed to borrow a SPA3102 with the latest firmware and have got it > to register using TLS, but I am still struggling with SRTP. ?Has > anyone managed to get SRTP working with the Linksys devices and if so, > can they direct me on how to do this. > > I have generated a mini-certificates and SRTP Private Key using the > gen-mc tool found at > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. > ?However, when ever I initiate a call from the SPA, I can see that the > call is not encrypted. > > Help appreciated. > > Thanks! ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From gkuri at ieee.org Thu Dec 3 14:17:25 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Thu, 3 Dec 2009 14:17:25 -0800 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> Message-ID: <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH. They do their proprietary Sipura key exchange only, not sure if Cisco plans on upgrading the firmware to ever support SDES on the ATAs. They added support for SDES to their IP Phones about 1 year ago, but nothing has happened with the ATAs as of yet. Gabe On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith wrote: > Hi All, > > I managed to borrow a SPA3102 with the latest firmware and have got it > to register using TLS, but I am still struggling with SRTP. ?Has > anyone managed to get SRTP working with the Linksys devices and if so, > can they direct me on how to do this. > > I have generated a mini-certificates and SRTP Private Key using the > gen-mc tool found at > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. > ?However, when ever I initiate a call from the SPA, I can see that the > call is not encrypted. > > Help appreciated. > > Thanks! > > > On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> Check out the Linksys SPA2102 >> >> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> wrote: >>> >>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>> doesn't seem to behave to well with TLS ... >>> >>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>> > Mark Campbell-Smith wrote: >>> >> Does the SPA3102 support TLS or only SRTP? >>> > >>> > I don't know, but supporting only SRTP would be ridiculous, since the >>> > keys >>> > would then be transmitted in the clear and therefore amenable to >>> > interception. >>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>> > secure. >>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>> > in >>> > RTP. >>> > >>> > I would be rather surprised were a hardware manufacturer to implement >>> > SRTP >>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>> > this >>> > forum that some manufacturers are really clueless... >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mcampbellsmith at gmail.com Thu Dec 3 14:34:29 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 09:34:29 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> Message-ID: <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange to appropriately support SRTP and FreeSWITCH I'll check with Cisco regarding their implementation then and try to find out when/if they will support standard SRTP encryption. So, back to my origianal question then. Are there any ATA's that support TLS AND SRTP with FreeSwitch? On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > exchange to appropriately support SRTP and FreeSWITCH. They do their > proprietary Sipura key exchange only, not sure if Cisco plans on > upgrading the firmware to ever support SDES on the ATAs. They added > support for SDES to their IP Phones about 1 year ago, but nothing has > happened with the ATAs as of yet. > > Gabe > > > On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > wrote: >> Hi All, >> >> I managed to borrow a SPA3102 with the latest firmware and have got it >> to register using TLS, but I am still struggling with SRTP. ?Has >> anyone managed to get SRTP working with the Linksys devices and if so, >> can they direct me on how to do this. >> >> I have generated a mini-certificates and SRTP Private Key using the >> gen-mc tool found at >> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >> ?However, when ever I initiate a call from the SPA, I can see that the >> call is not encrypted. >> >> Help appreciated. >> >> Thanks! >> >> >> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>> Check out the Linksys SPA2102 >>> >>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>> wrote: >>>> >>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>> doesn't seem to behave to well with TLS ... >>>> >>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>> > Mark Campbell-Smith wrote: >>>> >> Does the SPA3102 support TLS or only SRTP? >>>> > >>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>> > keys >>>> > would then be transmitted in the clear and therefore amenable to >>>> > interception. >>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>> > secure. >>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>> > in >>>> > RTP. >>>> > >>>> > I would be rather surprised were a hardware manufacturer to implement >>>> > SRTP >>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>> > this >>>> > forum that some manufacturers are really clueless... >>>> > >>>> > >>>> > _______________________________________________ >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mouncifbb at gmail.com Thu Dec 3 14:33:58 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Thu, 3 Dec 2009 17:33:58 -0500 Subject: [Freeswitch-users] Generate cdrs Message-ID: is it possible to run a javascript at the end of dialplan to generate cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file on machine reboots or shutdown signals. javascript or LUA for preferences? thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/30db44f7/attachment-0002.html From gkuri at ieee.org Thu Dec 3 15:25:29 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Thu, 3 Dec 2009 15:25:29 -0800 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911220121k5b0a0438udae727e09b8e986f@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> Message-ID: <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the Grandstream and Mediatrix devices (although I've never tried either one with FreeSWITCH). I've personally never had any good experience with the Grandstream ATAs. The Mediatrix ATAs are OK devices, but I've never personally tested them with SRTP w/SDES and FreeSWITCH, but supposedly they support it (so says their marketing material and docs). I'd see if Cisco has any plans to add support for it to the ATAs. Next time I see our Cisco SE, I'll try to poke him about it. Gabe On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith wrote: > Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > to appropriately support SRTP and FreeSWITCH > > I'll check with Cisco regarding their implementation then and try to > find out when/if they will support standard SRTP encryption. > > > So, back to my origianal question then. ?Are there any ATA's that > support TLS AND SRTP with FreeSwitch? > > > On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> exchange to appropriately support SRTP and FreeSWITCH. They do their >> proprietary Sipura key exchange only, not sure if Cisco plans on >> upgrading the firmware to ever support SDES on the ATAs. They added >> support for SDES to their IP Phones about 1 year ago, but nothing has >> happened with the ATAs as of yet. >> >> Gabe >> >> >> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> wrote: >>> Hi All, >>> >>> I managed to borrow a SPA3102 with the latest firmware and have got it >>> to register using TLS, but I am still struggling with SRTP. ?Has >>> anyone managed to get SRTP working with the Linksys devices and if so, >>> can they direct me on how to do this. >>> >>> I have generated a mini-certificates and SRTP Private Key using the >>> gen-mc tool found at >>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >>> ?However, when ever I initiate a call from the SPA, I can see that the >>> call is not encrypted. >>> >>> Help appreciated. >>> >>> Thanks! >>> >>> >>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>>> Check out the Linksys SPA2102 >>>> >>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>>> wrote: >>>>> >>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>>> doesn't seem to behave to well with TLS ... >>>>> >>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>>> > Mark Campbell-Smith wrote: >>>>> >> Does the SPA3102 support TLS or only SRTP? >>>>> > >>>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>>> > keys >>>>> > would then be transmitted in the clear and therefore amenable to >>>>> > interception. >>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>>> > secure. >>>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>>> > in >>>>> > RTP. >>>>> > >>>>> > I would be rather surprised were a hardware manufacturer to implement >>>>> > SRTP >>>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>>> > this >>>>> > forum that some manufacturers are really clueless... >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From oscav at hotmail.fr Thu Dec 3 15:33:57 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 3 Dec 2009 15:33:57 -0800 (PST) Subject: [Freeswitch-users] How to run a JS script periodically In-Reply-To: References: <26625147.post@talk.nabble.com> Message-ID: <26635167.post@talk.nabble.com> fs_cli looks like a good idea. I will try that. Many thanks Rob Rob Forman wrote: > > What about cron? > > Create a cron entry like: > */5 * * * * /usr/local/freeswitch/bin/fs_cli -x "jsrun yourscript &app()" > > But if you're just dumping global variables, you could easily retrieve > them > directly from fs_cli without running an app and process the output however > you'd like: > > /usr/local/freeswitch/bin/fs_cli -x "global_getvar" > > > On Thu, Dec 3, 2009 at 6:21 AM, Oscav wrote: > >> >> Hi, >> >> Someone knows how to run periodically a JS script ?? The purpose is to >> write >> to a db some global informations (Global Variables) about FS like every 5 >> minutes. >> >> Thanks. >> >> >> -- >> View this message in context: >> http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26635167.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Dec 3 15:49:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Dec 2009 17:49:20 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26633739.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> Message-ID: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Sigh, You just took it up a notch in terms of disdain and sarcasm. Why do people always only apologize sarcastically? I asked you to try the -hp and turn off the monotonic clock just to gather the results to help you. You completely missed it and just went on about the threads. Please save the "ok fine the code is perfect, blah blah" if you would have just read the email and answered the question I might have cared more about the status of your problem. I told you both of those threads need to be on their toes because they try to balance between a certian number of sql stmts or 500ms whatever comes first. When there are thousands of events per second being turned into SQL statements which are in turn compiled into large sql transactions. If you want to come up with a way that they can sleep longer until there is a sign of activity and stay busy for a few seconds then slow down again, that's probably possible but the process is already idle at 0% cpu so maybe you can appreciate why we are not rushing to work on it. Maybe I'll give it a go just to show you it has nothing to do with your problem. Please don't mock our comment about several years. You have no idea how hard this code was to develop and it's truly insulting. Its clear to see you are locked into assuming that the busy threads that are not all that busy because they are constantly yielding to the scheduler is breaking the timing code. I begged you to understand me when i told you that the err is not normal, most boxes do not see it doing nothing and there has to be a specific problem on your box or configuration. So instead of working with us you want to escalate to snotty comments. That's pretty normal on the internet I guess..... If you want to have a constructive conversation about our core, install FS on a normal box, use it for a few weeks, figure out everything about how it works then try.... There was pure speculation and conjecture in your original emails and I never said a word about it until you kept pushing. Kristian mentioned he never sees that on that same hardware did you even consider following up on why that is? I don't have your device, but I assume if you get it working well it will certainly help you more than it helps me so you could at least have the decency to believe what we are trying to tell you. On Thu, Dec 3, 2009 at 3:44 PM, eaf wrote: > > Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do > that. > At the moment, I hope it won't be necessary as I can make those "hyper" > threads behave, and will see how that goes first. I see where your > implementation could be coming from. There is a queue of SQL queries in > sofia.c processed by the worker thread. There are only two pop functions > available in APR: queue_pop() and queue_trypop(), so alas no option with a > timeout here. You don't want to block the thread in pop() indefinitely > because you chose that same worker needs to do ireg and gw processing once > in a while (separated by tens or hundreds of seconds, btw). You also want > to > be able to detect shutdown condition so that the worker doesn't hold up > profile thread. So you chose to poll for events every millisecond instead > of > just creating an apr_thread_cond_t for resource friendly signalling. > > I agree that the timer thread philosophy is great and was the right choice > for scaling, but I just don't comprehend responses to things like these > other SQL or sofia worker threads. Did somebody even remotely acknowledge > that busy loops at least in those areas that I showed may probably be a bad > idea and could've been eliminated? I've heard suggestions to bump up > priority, I've heard that the code was perfect already, that it's the > result > of 4-year effort, that I am arrogant, don't listen and don't understand > squat. > > I'm sorry if I gave you impression that I was looking for the bad parts in > the software. I apologized for that already. All I wanted was to have > constructive conversation, perhaps I'm not too good at it. Code is already > perfect according to you? Fine with me. > > > Anthony Minessale-2 wrote: > > > > no, > > > > I mean the one after that that you must have completely skipped with a > > command line option to try and a param to set in the config. It somewhat > > annoys me for taking the time to compose it now. I wrote all of the code > > you are talking about myself and I was trying to give you some > > suggestions.... > > > > Well, actually, you did answer my question about the platform so you > must > > have seen it..... > > > > The loops are not the cause of that migration message, something wrong > > with > > the hardware or the kernel is. > > Another guy just told you he does not see that problem on the same exact > > hardware. > > > > Even if you have a point about the sql threads, you could make a patch to > > slow them down but you cant slow down too much or you will not be able to > > handle 400 cps all asking to send updates to transactions in batches of > > thousands of sql stmts. Every line of that code is carefully designed so > > I > > don't know what else to tell you but to stop being so arrogant and > re-read > > this thread for all the advice you have totally ignored. I started out > > trying to help you but I have a lot of work to do. I thoroughly > explained > > it to you and you are choosing to ignore me so I guess I'm done. > > You can do whatever you want with your working copy, i'll see you in 3 or > > 4 > > years when you get up to speed with the rest of us........ > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/538b8cbc/attachment-0002.html From dujinfang at gmail.com Thu Dec 3 16:02:53 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 08:02:53 +0800 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: References: Message-ID: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> why not try mod_xml_cdr? 2009/12/4 Mouncif Benniane : > is it possible to run a javascript at the end of dialplan to generate cdrs? > because (mod_cdr_csv) is giving me hard time as it rotates Master file on > machine reboots or shutdown signals. > javascript or LUA for preferences? > > thank you > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Thu Dec 3 16:06:29 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 08:06:29 +0800 Subject: [Freeswitch-users] Cannot Do this Basic thing In-Reply-To: <4B180525.7060702@greatiam.com> References: <4B17F95F.2000108@greatiam.com> <87f2f3b90912030957l658d4f3alc6131aa585fff589@mail.gmail.com> <4B180525.7060702@greatiam.com> Message-ID: <23f91030912031606m4cc2698dyb53a2b22754d05ef@mail.gmail.com> You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ? 2009/12/4 Samuel Abekah-Mensah : > Hi > > Sorry .xm is a typo. I actually shut down the server and restarted. The > log says I need to create a domain of aaa.bbb.ccc.ddd (which is the > server IP address ) and then put the user in that domain. ?Isn't the > default domain that of the server FS is running on ? > 2319.xml is in /usr/local/freeswitch/conf/directory/default/ > > Thanks for your time > > > > Michael Collins wrote: >> >> >> On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah >> > wrote: >> >> ? ? I have copied 1001.xml in directory/default to a test user 2319.xm >> ? ? changing or instances of 1001 in the file to 2319. I then went into >> ? ? default.xml ?in directory folder and in one of the groups ?just >> ? ? mimicked >> ? ? 1001 details by changing 1001 to 2319. >> >> ? ? Connecting ?to FS gives Forbidden message. However 1001 connects >> ? ? without >> ? ? a problem. ?What have I missed ? >> >> ? ? Is there a place that just puts things in do this and that and that to >> ? ? create a new user ? >> >> >> Did you execute "reloadxml" from the fs cli before trying to connect >> with 2319? Also I'm assuming that "2319.xm" is a typo and you actually >> created "2319.xml" in the default/directory subdir. >> -MC >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From timuckun at gmail.com Thu Dec 3 16:22:53 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 4 Dec 2009 13:22:53 +1300 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <26625105.post@talk.nabble.com> References: <26625105.post@talk.nabble.com> Message-ID: <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> > > Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever) > ports fail being opened dynamically to work properly, or does SIP today > really work well over NAT firewalls? > Yes I get issues quite a bit with the server being behind a firewall. IAX is much nicer in this circumstance. From jason at jasonjgw.net Thu Dec 3 16:35:24 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 4 Dec 2009 11:35:24 +1100 Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> References: <26625105.post@talk.nabble.com> <855e4dcf0912031622p5bd32185m27e714957c8e9443@mail.gmail.com> Message-ID: <20091204003524.GA22701@jdc.jasonjgw.net> Tim Uckun wrote: > Yes I get issues quite a bit with the server being behind a firewall. > IAX is much nicer in this circumstance. I just set up an IPv6 over IPv4 tunnel and nat goes away. I have native IPv6 over ADSL now, as part of a trial that my ISP is conducting. As a result, one end of the conection doesn't go through a tunnel provider anymore. Given the problems I've had (and still have) with nat, I want to be rid of it as much as possible. Nevertheless, I agree that in a nat scenario, IAX can be easier to configure correctly. From mike at jerris.com Thu Dec 3 16:56:24 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Dec 2009 19:56:24 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> Message-ID: <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover. Mike On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: > On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: >> The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? >> > > From an as of now unkown SIP trunk provider (we are still in > negotiations with a couple of companies). > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Dec 3 17:47:35 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Dec 2009 20:47:35 -0500 Subject: [Freeswitch-users] Playing an rtp stream Message-ID: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> Hi there, It it possible do something like: Basically I have need to connect to incoming calls listen to an existing rtp stream - I know the IP and port. Any hints on achieving this would be much appreciated. Thanks Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091203/71b9ea52/attachment-0002.html From mcampbellsmith at gmail.com Thu Dec 3 18:26:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 13:26:17 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> Message-ID: <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Cheers Gabriel.. thanks for the information. I'll look at the Mediatrix ATA's as an alternative - has anyone had experience with those and TLS/SRTP? On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > Grandstream and Mediatrix devices (although I've never tried either > one with FreeSWITCH). > > I've personally never had any good experience with the Grandstream > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > support it (so says their marketing material and docs). > > I'd see if Cisco has any plans to add support for it to the ATAs. Next > time I see our Cisco SE, I'll try to poke him about it. > > Gabe > > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > wrote: >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange >> to appropriately support SRTP and FreeSWITCH >> >> I'll check with Cisco regarding their implementation then and try to >> find out when/if they will support standard SRTP encryption. >> >> >> So, back to my origianal question then. ?Are there any ATA's that >> support TLS AND SRTP with FreeSwitch? >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >>> exchange to appropriately support SRTP and FreeSWITCH. They do their >>> proprietary Sipura key exchange only, not sure if Cisco plans on >>> upgrading the firmware to ever support SDES on the ATAs. They added >>> support for SDES to their IP Phones about 1 year ago, but nothing has >>> happened with the ATAs as of yet. >>> >>> Gabe >>> >>> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >>> wrote: >>>> Hi All, >>>> >>>> I managed to borrow a SPA3102 with the latest firmware and have got it >>>> to register using TLS, but I am still struggling with SRTP. ?Has >>>> anyone managed to get SRTP working with the Linksys devices and if so, >>>> can they direct me on how to do this. >>>> >>>> I have generated a mini-certificates and SRTP Private Key using the >>>> gen-mc tool found at >>>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >>>> ?However, when ever I initiate a call from the SPA, I can see that the >>>> call is not encrypted. >>>> >>>> Help appreciated. >>>> >>>> Thanks! >>>> >>>> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >>>>> Check out the Linksys SPA2102 >>>>> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >>>>> wrote: >>>>>> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >>>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >>>>>> doesn't seem to behave to well with TLS ... >>>>>> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White wrote: >>>>>> > Mark Campbell-Smith wrote: >>>>>> >> Does the SPA3102 support TLS or only SRTP? >>>>>> > >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since the >>>>>> > keys >>>>>> > would then be transmitted in the clear and therefore amenable to >>>>>> > interception. >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to be >>>>>> > secure. >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works entirely >>>>>> > in >>>>>> > RTP. >>>>>> > >>>>>> > I would be rather surprised were a hardware manufacturer to implement >>>>>> > SRTP >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen often in >>>>>> > this >>>>>> > forum that some manufacturers are really clueless... >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Thu Dec 3 20:38:21 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 06:38:21 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911242346g674b7342v845066a117a2c773@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Message-ID: Hello, I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they should support TLS also (will try it next week; up to now I preffered to not use TLS so I can sniff the traffic and debug things). Regards, __Yehavi: 2009/12/4 Mark Campbell-Smith > Cheers Gabriel.. thanks for the information. > > I'll look at the Mediatrix ATA's as an alternative - has anyone had > experience with those and TLS/SRTP? > > > On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > > Grandstream and Mediatrix devices (although I've never tried either > > one with FreeSWITCH). > > > > I've personally never had any good experience with the Grandstream > > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > > support it (so says their marketing material and docs). > > > > I'd see if Cisco has any plans to add support for it to the ATAs. Next > > time I see our Cisco SE, I'll try to poke him about it. > > > > Gabe > > > > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > > wrote: > >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > >> to appropriately support SRTP and FreeSWITCH > >> > >> I'll check with Cisco regarding their implementation then and try to > >> find out when/if they will support standard SRTP encryption. > >> > >> > >> So, back to my origianal question then. Are there any ATA's that > >> support TLS AND SRTP with FreeSwitch? > >> > >> > >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > >>> exchange to appropriately support SRTP and FreeSWITCH. They do their > >>> proprietary Sipura key exchange only, not sure if Cisco plans on > >>> upgrading the firmware to ever support SDES on the ATAs. They added > >>> support for SDES to their IP Phones about 1 year ago, but nothing has > >>> happened with the ATAs as of yet. > >>> > >>> Gabe > >>> > >>> > >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > >>> wrote: > >>>> Hi All, > >>>> > >>>> I managed to borrow a SPA3102 with the latest firmware and have got it > >>>> to register using TLS, but I am still struggling with SRTP. Has > >>>> anyone managed to get SRTP working with the Linksys devices and if so, > >>>> can they direct me on how to do this. > >>>> > >>>> I have generated a mini-certificates and SRTP Private Key using the > >>>> gen-mc tool found at > >>>> > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 > . > >>>> However, when ever I initiate a call from the SPA, I can see that the > >>>> call is not encrypted. > >>>> > >>>> Help appreciated. > >>>> > >>>> Thanks! > >>>> > >>>> > >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > >>>>> Check out the Linksys SPA2102 > >>>>> > >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > >>>>> wrote: > >>>>>> > >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the > >>>>>> Grandstream HandyTone 503. But, again according to the wiki, that > >>>>>> doesn't seem to behave to well with TLS ... > >>>>>> > >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White > wrote: > >>>>>> > Mark Campbell-Smith wrote: > >>>>>> >> Does the SPA3102 support TLS or only SRTP? > >>>>>> > > >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since > the > >>>>>> > keys > >>>>>> > would then be transmitted in the clear and therefore amenable to > >>>>>> > interception. > >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to > be > >>>>>> > secure. > >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works > entirely > >>>>>> > in > >>>>>> > RTP. > >>>>>> > > >>>>>> > I would be rather surprised were a hardware manufacturer to > implement > >>>>>> > SRTP > >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen > often in > >>>>>> > this > >>>>>> > forum that some manufacturers are really clueless... > >>>>>> > > >>>>>> > > >>>>>> > _______________________________________________ > >>>>>> > FreeSWITCH-users mailing list > >>>>>> > FreeSWITCH-users at lists.freeswitch.org > >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> > http://www.freeswitch.org > >>>>>> > > >>>>>> > >>>>>> _______________________________________________ > >>>>>> FreeSWITCH-users mailing list > >>>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> FreeSWITCH-users mailing list > >>>>> FreeSWITCH-users at lists.freeswitch.org > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/6ed9e6a3/attachment-0002.html From yehavi.bourvine at gmail.com Thu Dec 3 20:40:25 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 06:40:25 +0200 Subject: [Freeswitch-users] Cisco IOS gateway: command to send connected line name In-Reply-To: References: <4B1800C7.7010800@metik.com> Message-ID: I am taking my words back... The Cisco sends back what I want. I got confused because the Nortel sends the name only for the connected PBX and not for the othes ones (although it gets this infomation from them). Thanks, __Yehavi: 2009/12/3 Yehavi Bourvine > Unfortunately this didn't help... Incoming calls from ISDN to SIP sends > back to ISDN the name of the destination, but not the other way around... > > Thanks! __Yehavi: > > 2009/12/3 Metik > > Yehavi, >> >> There are a few variations of transmitting this information... If you >> have already enabled a supplemental isdn service profile, try adding the >> following to the PRI you are using: >> >> (config-if)#isdn outgoing ie facility >> (config-if)#iisdn outgoing ie extended-facility >> (config-if)#isdn outgoing display-ie >> (config-if)#isdn outgoing ie caller-number >> (config-if)#isdn outgoing ie called-number >> >> -metik >> >> Yehavi Bourvine wrote: >> > Hello, >> > >> > We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On >> > the PRI there is a Nortel with Q.Sig. After a lot of configuration >> > trials I've managed to set it to send back the connected name over the >> > SIP (i.e. when a call goes from SIP to PRI, the PRI sends back the >> > connected name and then the Cisco adds it as a Remote-Party-ID). >> > However, I did not save it and a power outage cleared this config. In >> > my age I don't remember what I've done... >> > >> > Anyone knows the correct config? >> > >> > Thanks! __Yehavi: >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/70f2cadc/attachment-0002.html From mcampbellsmith at gmail.com Thu Dec 3 21:01:21 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 4 Dec 2009 16:01:21 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <20091125081453.GA28340@jdc.jasonjgw.net> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> Message-ID: <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Thanks Yehavi, I would be very interested to find out how your test goes... can you report back after you have tested it? Thanks! On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine wrote: > Hello, > > ? I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they > should support TLS also (will try it next week; up to now I preffered to not > use TLS so I can sniff the traffic and debug things). > > ???????????????? Regards, __Yehavi: > > 2009/12/4 Mark Campbell-Smith >> >> Cheers Gabriel.. thanks for the information. >> >> I'll look at the Mediatrix ATA's as an alternative - has anyone had >> experience with those and TLS/SRTP? >> >> >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the >> > Grandstream and Mediatrix devices (although I've never tried either >> > one with FreeSWITCH). >> > >> > I've personally never had any good experience with the Grandstream >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they >> > support it (so says their marketing material and docs). >> > >> > I'd see if Cisco has any plans to add support for it to the ATAs. Next >> > time I see our Cisco SE, I'll try to poke him about it. >> > >> > Gabe >> > >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith >> > wrote: >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange >> >> to appropriately support SRTP and FreeSWITCH >> >> >> >> I'll check with Cisco regarding their implementation then and try to >> >> find out when/if they will support standard SRTP encryption. >> >> >> >> >> >> So, back to my origianal question then. ?Are there any ATA's that >> >> support TLS AND SRTP with FreeSwitch? >> >> >> >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do their >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on >> >>> upgrading the firmware to ever support SDES on the ATAs. They added >> >>> support for SDES to their IP Phones about 1 year ago, but nothing has >> >>> happened with the ATAs as of yet. >> >>> >> >>> Gabe >> >>> >> >>> >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> >>> wrote: >> >>>> Hi All, >> >>>> >> >>>> I managed to borrow a SPA3102 with the latest firmware and have got >> >>>> it >> >>>> to register using TLS, but I am still struggling with SRTP. ?Has >> >>>> anyone managed to get SRTP working with the Linksys devices and if >> >>>> so, >> >>>> can they direct me on how to do this. >> >>>> >> >>>> I have generated a mini-certificates and SRTP Private Key using the >> >>>> gen-mc tool found at >> >>>> >> >>>> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3. >> >>>> ?However, when ever I initiate a call from the SPA, I can see that >> >>>> the >> >>>> call is not encrypted. >> >>>> >> >>>> Help appreciated. >> >>>> >> >>>> Thanks! >> >>>> >> >>>> >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> >>>>> Check out the Linksys SPA2102 >> >>>>> >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> >>>>> wrote: >> >>>>>> >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> >>>>>> Grandstream HandyTone 503. ?But, again according to the wiki, that >> >>>>>> doesn't seem to behave to well with TLS ... >> >>>>>> >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White >> >>>>>> wrote: >> >>>>>> > Mark Campbell-Smith wrote: >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? >> >>>>>> > >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, since >> >>>>>> > the >> >>>>>> > keys >> >>>>>> > would then be transmitted in the clear and therefore amenable to >> >>>>>> > interception. >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order to >> >>>>>> > be >> >>>>>> > secure. >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works >> >>>>>> > entirely >> >>>>>> > in >> >>>>>> > RTP. >> >>>>>> > >> >>>>>> > I would be rather surprised were a hardware manufacturer to >> >>>>>> > implement >> >>>>>> > SRTP >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen >> >>>>>> > often in >> >>>>>> > this >> >>>>>> > forum that some manufacturers are really clueless... >> >>>>>> > >> >>>>>> > >> >>>>>> > _______________________________________________ >> >>>>>> > FreeSWITCH-users mailing list >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>> > >> >>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>> > http://www.freeswitch.org >> >>>>>> > >> >>>>>> >> >>>>>> _______________________________________________ >> >>>>>> FreeSWITCH-users mailing list >> >>>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>>> >> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>>> http://www.freeswitch.org >> >>>>> >> >>>>> >> >>>>> _______________________________________________ >> >>>>> FreeSWITCH-users mailing list >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>>> http://www.freeswitch.org >> >>>>> >> >>>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> >> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>>> >> >>> >> >>> _______________________________________________ >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From neilp at cs.stanford.edu Thu Dec 3 22:28:50 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 11:58:50 +0530 Subject: [Freeswitch-users] record mp3s Message-ID: Hi All, This is a great list, thanks for all of the support! For my IVR app running on FS, we we accept potentially long audio recordings. Is it possible (in lua) to save recorded as mp3? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/0d1233ec/attachment-0002.html From mrene_lists at avgs.ca Thu Dec 3 22:31:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 01:31:12 -0500 Subject: [Freeswitch-users] record mp3s In-Reply-To: References: Message-ID: <8FADAED2-E609-4F01-B81F-010E242A9F0A@avgs.ca> Hi Neil, If you have mod_shout loaded and use a .mp3 file as you recording filename, it'll automagically encode it. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 1:28 AM, Neil Patel wrote: > Hi All, > > This is a great list, thanks for all of the support! > > For my IVR app running on FS, we we accept potentially long audio > recordings. Is it possible (in lua) to save recorded as mp3? > > Thanks, > Neil > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From neilp at cs.stanford.edu Thu Dec 3 22:36:04 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 12:06:04 +0530 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: Thanks all for your help. I got around this by running ./Setup and installing wanpipe in TDM API mode (it says it's the default for FS). I then uncommented the mod_openzap line in modules.conf when installing FS. Finally I ran wancfg_fs which creates appropriate config files for you for your FS installation. I believe openzap is now installed properly: 2009-12-04 12:04:52.411017 [INFO] zap_io.c:2451 Loading IO from /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] 2009-12-04 12:04:52.411126 [INFO] zap_io.c:2251 auto-loaded 'wanpipe' 2009-12-04 12:04:52.411311 [INFO] ozmod_wanpipe.c:287 configuring device s1c1 as OpenZAP device 1:1 fd:14 DTMF: software 2009-12-04 12:04:52.411377 [INFO] ozmod_wanpipe.c:287 configuring device s1c2 as OpenZAP device 1:2 fd:15 DTMF: software 2009-12-04 12:04:52.411444 [INFO] ozmod_wanpipe.c:287 configuring device s1c3 as OpenZAP device 1:3 fd:17 DTMF: software 2009-12-04 12:04:52.411509 [INFO] ozmod_wanpipe.c:287 configuring device s1c4 as OpenZAP device 1:4 fd:18 DTMF: software 2009-12-04 12:04:52.411575 [INFO] ozmod_wanpipe.c:287 configuring device s1c5 as OpenZAP device 1:5 fd:19 DTMF: software 2009-12-04 12:04:52.411639 [INFO] ozmod_wanpipe.c:287 configuring device s1c6 as OpenZAP device 1:6 fd:20 DTMF: software 2009-12-04 12:04:52.411707 [INFO] ozmod_wanpipe.c:287 configuring device s1c7 as OpenZAP device 1:7 fd:21 DTMF: software 2009-12-04 12:04:52.411771 [INFO] ozmod_wanpipe.c:287 configuring device s1c8 as OpenZAP device 1:8 fd:22 DTMF: software 2009-12-04 12:04:52.411837 [INFO] ozmod_wanpipe.c:287 configuring device s1c9 as OpenZAP device 1:9 fd:23 DTMF: software 2009-12-04 12:04:52.411903 [INFO] ozmod_wanpipe.c:287 configuring device s1c10 as OpenZAP device 1:10 fd:24 DTMF: software 2009-12-04 12:04:52.411969 [INFO] ozmod_wanpipe.c:287 configuring device s1c11 as OpenZAP device 1:11 fd:25 DTMF: software 2009-12-04 12:04:52.412034 [INFO] ozmod_wanpipe.c:287 configuring device s1c12 as OpenZAP device 1:12 fd:26 DTMF: software 2009-12-04 12:04:52.412102 [INFO] ozmod_wanpipe.c:287 configuring device s1c13 as OpenZAP device 1:13 fd:27 DTMF: software 2009-12-04 12:04:52.412179 [INFO] ozmod_wanpipe.c:287 configuring device s1c14 as OpenZAP device 1:14 fd:28 DTMF: software 2009-12-04 12:04:52.412244 [INFO] ozmod_wanpipe.c:287 configuring device s1c15 as OpenZAP device 1:15 fd:29 DTMF: software TDM API: CMD: 18 : Operation not supported 2009-12-04 12:04:52.412416 [INFO] ozmod_wanpipe.c:287 configuring device s1c16 as OpenZAP device 1:16 fd:30 DTMF: none 2009-12-04 12:04:52.412503 [INFO] ozmod_wanpipe.c:287 configuring device s1c17 as OpenZAP device 1:17 fd:31 DTMF: software 2009-12-04 12:04:52.412568 [INFO] ozmod_wanpipe.c:287 configuring device s1c18 as OpenZAP device 1:18 fd:32 DTMF: software 2009-12-04 12:04:52.412634 [INFO] ozmod_wanpipe.c:287 configuring device s1c19 as OpenZAP device 1:19 fd:33 DTMF: software 2009-12-04 12:04:52.412708 [INFO] ozmod_wanpipe.c:287 configuring device s1c20 as OpenZAP device 1:20 fd:34 DTMF: software 2009-12-04 12:04:52.412771 [INFO] ozmod_wanpipe.c:287 configuring device s1c21 as OpenZAP device 1:21 fd:35 DTMF: software 2009-12-04 12:04:52.412838 [INFO] ozmod_wanpipe.c:287 configuring device s1c22 as OpenZAP device 1:22 fd:36 DTMF: software 2009-12-04 12:04:52.412902 [INFO] ozmod_wanpipe.c:287 configuring device s1c23 as OpenZAP device 1:23 fd:37 DTMF: software 2009-12-04 12:04:52.412948 [INFO] ozmod_wanpipe.c:287 configuring device s1c24 as OpenZAP device 1:24 fd:38 DTMF: software 2009-12-04 12:04:52.412988 [INFO] ozmod_wanpipe.c:287 configuring device s1c25 as OpenZAP device 1:25 fd:39 DTMF: software 2009-12-04 12:04:52.413018 [INFO] ozmod_wanpipe.c:287 configuring device s1c26 as OpenZAP device 1:26 fd:40 DTMF: software 2009-12-04 12:04:52.413041 [INFO] ozmod_wanpipe.c:287 configuring device s1c27 as OpenZAP device 1:27 fd:41 DTMF: software 2009-12-04 12:04:52.413063 [INFO] ozmod_wanpipe.c:287 configuring device s1c28 as OpenZAP device 1:28 fd:42 DTMF: software 2009-12-04 12:04:52.413086 [INFO] ozmod_wanpipe.c:287 configuring device s1c29 as OpenZAP device 1:29 fd:43 DTMF: software 2009-12-04 12:04:52.413106 [INFO] ozmod_wanpipe.c:287 configuring device s1c30 as OpenZAP device 1:30 fd:44 DTMF: software 2009-12-04 12:04:52.413128 [INFO] ozmod_wanpipe.c:287 configuring device s1c31 as OpenZAP device 1:31 fd:45 DTMF: software 2009-12-04 12:04:52.413142 [INFO] zap_io.c:2374 Configured 31 channel(s) 2009-12-04 12:04:52.431405 [INFO] zap_io.c:2468 Loading SIG from /usr/local/freeswitch/mod/ozmod_ss7_boost.so 2009-12-04 12:04:52.431441 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost' 2009-12-04 12:04:52.431541 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_openzap] 2009-12-04 12:04:52.431553 [NOTICE] switch_loadable_module.c:142 Adding Endpoint 'openzap' 2009-12-04 12:04:52.431638 [NOTICE] switch_loadable_module.c:248 Adding Application 'disable_ec' 2009-12-04 12:04:52.431659 [NOTICE] switch_loadable_module.c:270 Adding API Function 'oz' 2009-12-04 12:04:52.432009 [WARNING] ss7_boost_client.c:244 TX EVENT (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0 Does this look right? Thanks. On Mon, Nov 30, 2009 at 9:09 PM, Moises Silva wrote: > On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: > >> Hi All, >> >> I am currently installing a Sangoma A102 card to work with FS using >> wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get >> openzap-related modules to compile: >> >> > cd wanpipe-3.5.6.5/ >> > make openzap >> ... >> make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >> make -C api/libstelephony clean >> make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >> make[1]: *** No rule to make target `clean'. Stop. >> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >> make: *** [all_lib] Error 2 >> >> The libstelephony directory has no Makefile in it. Why is it missing? Is >> there a version of wanpipe drivers that will work? I have been unsuccessful >> with 3.4.4 and 3.5.6 in similar fashion. >> >> > Hi Neil, > > Most likely the creation of the Makefile failed (since you mention you > can't see a Makefile). Please be sure to have installed the pre-requisites > listed at http://wiki.sangoma.com/Requirements > > Particularly in this case, libtool, autoconf and automake packages. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bf531d7a/attachment-0002.html From jbr at consiglia.dk Thu Dec 3 22:40:22 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 4 Dec 2009 07:40:22 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: Anthony, you advised me to use MySQL as the core database in order to access it from Lua. I'm testing that as a work-around. Still, I guess that your choice of SQLite as the default core database have been taken from efficiency or stability considerations. Using MySQL through an ODBC-connector does not sound as a clean solution. Have you any experience about "how bad" it is to use the ODBC MySQL combination in terms of stability, memory leaks and efficiency? Regards Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/333d0202/attachment-0002.html From mrene_lists at avgs.ca Thu Dec 3 22:42:34 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 01:42:34 -0500 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <3E4F23C1-9835-497C-B587-645E4B53F043@avgs.ca> ODBC isnt as bad as its used to be. We use it with postgresql every day and are very happy with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 1:40 AM, Jon Bruel wrote: > Anthony, you advised me to use MySQL as the core database in order > to access it from Lua. I?m testing that as a work-around. > > Still, I guess that your choice of SQLite as the default core > database have been taken from efficiency or stability > considerations. Using MySQL through an ODBC-connector does not sound > as a clean solution. Have you any experience about ?how bad? it is > to use the ODBC MySQL combination in terms of stability, memory > leaks and efficiency? > > Regards > > Jon Br?el > Consiglia Telecommunications > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > CVR: 27047882 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/60d2305a/attachment-0002.html From yehavi.bourvine at gmail.com Fri Dec 4 01:19:51 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 4 Dec 2009 11:19:51 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30911250034n4ce80e6bned28a11fdcd6a7d1@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: I'll report when I am done. So far I've enabled only SRTP and both support it. __Yehavi: 2009/12/4 Mark Campbell-Smith > Thanks Yehavi, > > I would be very interested to find out how your test goes... can you > report back after you have tested it? > > Thanks! > > On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine > wrote: > > Hello, > > > > I have AudioCodes MP and Vega ATA adapters. They both support SRTP; > they > > should support TLS also (will try it next week; up to now I preffered to > not > > use TLS so I can sniff the traffic and debug things). > > > > Regards, __Yehavi: > > > > 2009/12/4 Mark Campbell-Smith > >> > >> Cheers Gabriel.. thanks for the information. > >> > >> I'll look at the Mediatrix ATA's as an alternative - has anyone had > >> experience with those and TLS/SRTP? > >> > >> > >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: > >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the > >> > Grandstream and Mediatrix devices (although I've never tried either > >> > one with FreeSWITCH). > >> > > >> > I've personally never had any good experience with the Grandstream > >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally > >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they > >> > support it (so says their marketing material and docs). > >> > > >> > I'd see if Cisco has any plans to add support for it to the ATAs. Next > >> > time I see our Cisco SE, I'll try to poke him about it. > >> > > >> > Gabe > >> > > >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith > >> > wrote: > >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange > >> >> to appropriately support SRTP and FreeSWITCH > >> >> > >> >> I'll check with Cisco regarding their implementation then and try to > >> >> find out when/if they will support standard SRTP encryption. > >> >> > >> >> > >> >> So, back to my origianal question then. Are there any ATA's that > >> >> support TLS AND SRTP with FreeSwitch? > >> >> > >> >> > >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri wrote: > >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key > >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do their > >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on > >> >>> upgrading the firmware to ever support SDES on the ATAs. They added > >> >>> support for SDES to their IP Phones about 1 year ago, but nothing > has > >> >>> happened with the ATAs as of yet. > >> >>> > >> >>> Gabe > >> >>> > >> >>> > >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith > >> >>> wrote: > >> >>>> Hi All, > >> >>>> > >> >>>> I managed to borrow a SPA3102 with the latest firmware and have got > >> >>>> it > >> >>>> to register using TLS, but I am still struggling with SRTP. Has > >> >>>> anyone managed to get SRTP working with the Linksys devices and if > >> >>>> so, > >> >>>> can they direct me on how to do this. > >> >>>> > >> >>>> I have generated a mini-certificates and SRTP Private Key using the > >> >>>> gen-mc tool found at > >> >>>> > >> >>>> > http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 > . > >> >>>> However, when ever I initiate a call from the SPA, I can see that > >> >>>> the > >> >>>> call is not encrypted. > >> >>>> > >> >>>> Help appreciated. > >> >>>> > >> >>>> Thanks! > >> >>>> > >> >>>> > >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: > >> >>>>> Check out the Linksys SPA2102 > >> >>>>> > >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith > >> >>>>> wrote: > >> >>>>>> > >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the > >> >>>>>> Grandstream HandyTone 503. But, again according to the wiki, > that > >> >>>>>> doesn't seem to behave to well with TLS ... > >> >>>>>> > >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White > > >> >>>>>> wrote: > >> >>>>>> > Mark Campbell-Smith wrote: > >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? > >> >>>>>> > > >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, > since > >> >>>>>> > the > >> >>>>>> > keys > >> >>>>>> > would then be transmitted in the clear and therefore amenable > to > >> >>>>>> > interception. > >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order > to > >> >>>>>> > be > >> >>>>>> > secure. > >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it works > >> >>>>>> > entirely > >> >>>>>> > in > >> >>>>>> > RTP. > >> >>>>>> > > >> >>>>>> > I would be rather surprised were a hardware manufacturer to > >> >>>>>> > implement > >> >>>>>> > SRTP > >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen > >> >>>>>> > often in > >> >>>>>> > this > >> >>>>>> > forum that some manufacturers are really clueless... > >> >>>>>> > > >> >>>>>> > > >> >>>>>> > _______________________________________________ > >> >>>>>> > FreeSWITCH-users mailing list > >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org > >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>> > > >> >>>>>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>> > http://www.freeswitch.org > >> >>>>>> > > >> >>>>>> > >> >>>>>> _______________________________________________ > >> >>>>>> FreeSWITCH-users mailing list > >> >>>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>>> > >> >>>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>>> http://www.freeswitch.org > >> >>>>> > >> >>>>> > >> >>>>> _______________________________________________ > >> >>>>> FreeSWITCH-users mailing list > >> >>>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>>> > >> >>>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>>> http://www.freeswitch.org > >> >>>>> > >> >>>>> > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at lists.freeswitch.org > >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>>> > >> >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>>> http://www.freeswitch.org > >> >>>> > >> >>> > >> >>> _______________________________________________ > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >>> > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bc9e1245/attachment-0002.html From jbr at consiglia.dk Fri Dec 4 01:24:01 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 4 Dec 2009 10:24:01 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I have now tested the FS with core db configured using MySql (by modifying the switch.conf.xml file). Unfortunately, it does not solve my problem because some of the core tables still remain as active SQLite tables. After restarting the FS in the new configuration (with SQLite database core deleted), the following tables are created in MySql and SQLite: MySQL: aliases, complete, nat and tasks (database starting with no tables prior to FS restart). SQLite: aliases, calls, channels, interfaces, nat and tasks. As I would like to access the channels table using Lua, the change did not fix my problem. I have positive verified that the channels table is active and populated during calls. Are there other places where I should define the usage of the MySql database? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/99e4251a/attachment-0002.html From codecomplete at free.fr Fri Dec 4 02:32:14 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 4 Dec 2009 02:32:14 -0800 (PST) Subject: [Freeswitch-users] IAX? Issues connecting road warriors with SIP? In-Reply-To: <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> References: <26625105.post@talk.nabble.com> <071E8C81-A5A5-402E-8E1B-A891028E4A21@jerris.com> Message-ID: <26635842.post@talk.nabble.com> Michael Jerris wrote: > with a client that does not support stun or at least rfc 3581 the results > are much more sketchy and require more hacks on the server side, but with > enough effort can almost always be made to work. Thanks Mike for the feedback. If a user has a problem using my FS server, I'll check what client they have. For those interested, here's what RFC 3581 adds to SIP: "Session Initiation Protocol (SIP) operates over UDP and TCP, among others. When used with UDP, responses to requests are returned to the source address the request came from, and to the port written into the topmost Via header field value of the request. This behavior is not desirable in many cases, most notably, when the client is behind a Network Address Translator (NAT). This extension defines a new parameter for the Via header field, called "rport", that allows a client to request that the server send the response back to the source IP address and port from which the request originated." -- View this message in context: http://old.nabble.com/IAX--Issues-connecting-road-warriors-with-SIP--tp26625105p26635842.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Fri Dec 4 04:39:10 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 4 Dec 2009 13:39:10 +0100 Subject: [Freeswitch-users] HA questions. In-Reply-To: <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> <55278139-0749-4102-ABEB-0E9F579D628E@jerris.com> <855e4dcf0912031240w3a715444j1fbee082c7fbf39e@mail.gmail.com> <6A35F881-7E15-4E92-B259-CD0C5493A5EB@jerris.com> Message-ID: <65d96fc80912040439x727f2a0jdb6bd07a41971f0@mail.gmail.com> Hi Mike, Lets suppose we have: - 2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. ( http://www.ultramonkey.org/3/topologies/ha-overview.html) - freeswitch installed on every machine configured to use mysql in the core via odbc - both freeswitch have identical dialplan and directory configuration - mysql installed on every machine (with replication between the DBs) - SIP Trunks towards the upper provider (without registration but i should work with registration) - SIP Phones/Terminals registering to the active freeswitch When a terminal registers to the active freeswitch, the registration is propagated to the inactive one via DB replication. Now, lets suppose we have a switchover ... of course we will lose the ongoing calls but new calls (from SIP Phones) should be able to establish. The same applies to incoming calls from the upper provider. Im just talking about HA here not loadbalancing and performance scaling... what do you think about that? On Fri, Dec 4, 2009 at 1:56 AM, Michael Jerris wrote: > so your registering to the provider to get the calls? If so, this gets > tricky, the provider likely does not support multiple registrations, even if > they did they probably send the call to both registered endpoints. With > this big unknown its not very easy to suggest a good solution. If I were > looking to set this up without needing proxies I would want to use srv > records and naptr records and a provider that would balance using these > including failiover. > > Mike > > > On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: > > > On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris wrote: > >> The easiest place to do this is at the point you send the calls to > FreeSWITCH. How are the calls coming in? > >> > > > > From an as of now unkown SIP trunk provider (we are still in > > negotiations with a couple of companies). > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/ca6d2e10/attachment-0002.html From mouncifbb at gmail.com Fri Dec 4 06:33:53 2009 From: mouncifbb at gmail.com (Mouncifbb) Date: Fri, 4 Dec 2009 09:33:53 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> Message-ID: <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> I don't want to use XML cdr it puts each call on individual files so is it possible to include a JavaScript at the end of dialplan to collect info about the session? Thanks On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > why not try mod_xml_cdr? > > 2009/12/4 Mouncif Benniane : >> is it possible to run a javascript at the end of dialplan to >> generate cdrs? >> because (mod_cdr_csv) is giving me hard time as it rotates Master >> file on >> machine reboots or shutdown signals. >> javascript or LUA for preferences? >> >> thank you >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Fri Dec 4 06:59:27 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Dec 2009 22:59:27 +0800 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> Message-ID: <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> 2009/12/4 Mouncifbb : > I don't want to use XML cdr it puts each call on individual files so It posts to a http server, and fall back to a xml file if server fails > is it possible to include a JavaScript at the end of dialplan to > collect info about the session? > I think the answer is yes but where would you store the collected info? > Thanks > > > On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > >> why not try mod_xml_cdr? >> >> 2009/12/4 Mouncif Benniane : >>> is it possible to run a javascript at the end of dialplan to >>> generate cdrs? >>> because (mod_cdr_csv) is giving me hard time as it rotates Master >>> file on >>> machine reboots or shutdown signals. >>> javascript or LUA for preferences? >>> >>> thank you >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mouncifbb at gmail.com Fri Dec 4 07:14:24 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Fri, 4 Dec 2009 10:14:24 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> References: <23f91030912031602u6c5b6ed1k1c49c732bc93e3b9@mail.gmail.com> <191A139D-B74C-40B2-A1EA-0875000D79FE@gmail.com> <23f91030912040659q37c29185y191daf161c207775@mail.gmail.com> Message-ID: I wanna store it on different file out of cdr-csv directory, basically making another copy of the Master.csv cdr file and also because I couldn't trust whether the Master.csv will be rotated accidentally again. Thanks On Fri, Dec 4, 2009 at 9:59 AM, Seven Du wrote: > 2009/12/4 Mouncifbb : > > I don't want to use XML cdr it puts each call on individual files so > > It posts to a http server, and fall back to a xml file if server fails > > > is it possible to include a JavaScript at the end of dialplan to > > collect info about the session? > > > > I think the answer is yes but where would you store the collected info? > > > Thanks > > > > > > On Dec 3, 2009, at 7:02 PM, Seven Du wrote: > > > >> why not try mod_xml_cdr? > >> > >> 2009/12/4 Mouncif Benniane : > >>> is it possible to run a javascript at the end of dialplan to > >>> generate cdrs? > >>> because (mod_cdr_csv) is giving me hard time as it rotates Master > >>> file on > >>> machine reboots or shutdown signals. > >>> javascript or LUA for preferences? > >>> > >>> thank you > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >>> users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/2ee358cd/attachment-0002.html From freeswitch-users-list at metik.com Fri Dec 4 07:24:36 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 04 Dec 2009 10:24:36 -0500 Subject: [Freeswitch-users] HA questions. In-Reply-To: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> References: <855e4dcf0912021649r6a7afa0fn50fa8510e65210c2@mail.gmail.com> Message-ID: <4B1929B4.5060507@metik.com> Since you seem to have most of the heavy lifting squared away with FS (e.g. database replication) and before reinventing the wheel, I would recommend that you speak to a few VoIP providers and see if they will do this for you as part of your service. Those that are using carrier class platforms (so-called active clustering) should be able to do this without too much effort on their part. If you reach any dead ends, please feel free to contact me off list. The one thing that you may want to keep in mind is that unless FS is not involved in the media flow (or has a chance to redirect the call to another FS), existing calls will be dropped. FS has no mechanism for exchanging/mirroring stateful signaling and media information between other FS nodes to specifically facilitate failover. As I believe the developers have indicated in this list, to do so would require significant investments in time and resources to implement it in the sofia sip stack at the moment. -metik Tim Uckun wrote: > I have read some of the archived emails about HA, loadbalancing, > failover etc and I am still a bit confused about how I could set up > some sort of resiliency with freeswitch. > > My situation is much less complex than the scenarios people were > talking about and I hoping the solution is similarly much less > complex. > > I have two machines. Both will run freeswitch and also an IVR > application with local databases. I will take care of the database, > application and configuration synchronization between the two > machines. Ideally the calls would be load balanced between the > machines and if any application falls down then the calls should go to > the other machine. Same if I take a machine down for whatever reason. > > If a machine goes down I am willing to "lose" those people who were > making a call at the time. I do have a flag in the application which > will stop answering the calls while processing the existing calls for > a graceful shutdown and hopefully the load balancer would shuttle the > calls to the other machine while this is happening. > > At this stage everything is done via SIP. > > My questions are... > > Do I have to have a sip proxy? If the answer is yes it seems like I > have to set up two sip proxies so I don't have another single point of > failure. Can I load the sip proxies on the same machine? Do I need two > more machines? > > If I take load balancing out of the picture would it be possible to do > a simple linux HA or a windows built in ip failover solution? Would a > simple IP failover work over UDP or would I have to use IAX and tcp/ip > ? > > Is it better to go the virtualization route? > > Sorry if these are dumb questions. I am just trying to get my head > wrapped around this. I don't need five nines (although that would be > awesome), I just want a reasonable degree of assurance that my app can > keep taking calls in case something weird happens. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Dec 4 07:38:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 09:38:59 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912040738r51056421t6122b3b408bc1be2@mail.gmail.com> That means you mysql is not configured to do transactions so it failed over back to sqlite. if you scan for the warning message you will see the option you have to set and you may possibly have to update your myodbc odbc driver. To answer you other question about the sqlite, like I said the lua does not have the object coded like js does so it would be a project to implement it. You can also consider using ODBC plugin for lua to access the sqlite. On Fri, Dec 4, 2009 at 3:24 AM, Jon Bruel wrote: > I have now tested the FS with core db configured using MySql (by > modifying the switch.conf.xml file). Unfortunately, it does not solve my > problem because some of the core tables still remain as active SQLite > tables. > > > > After restarting the FS in the new configuration (with SQLite database core > deleted), the following tables are created in MySql and SQLite: > > > > MySQL: aliases, complete, nat and tasks (database starting with no tables > prior to FS restart). > > SQLite: aliases, calls, channels, interfaces, nat and tasks. > > > > As I would like to access the channels table using Lua, the change did not > fix my problem. I have positive verified that the channels table is active > and populated during calls. > > > > Are there other places where I should define the usage of the MySql > database? > > > > > > *Jon Br?el* > Consiglia Telecommunications > > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > > CVR: 27047882 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/88f41a3b/attachment-0002.html From neilp at cs.stanford.edu Fri Dec 4 07:58:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 4 Dec 2009 21:28:58 +0530 Subject: [Freeswitch-users] IVR apps in lua Message-ID: Hi All, I haven't found a substantial example of IVR applications implemented in lua. Can anyone suggest where to look? My issue has to do with appropriate coding style. I am implementing a voice message board application in lua. I want to allow the user to dial buttons to navigate forward and back in the list of messages. One way to implement playmessage() is to check for a forward/back command while playing the current message, and if a command is given to invoke playmessage() with the prev/next message in the list. However, this leaves a chain of unreturned playmessage calls on the execution stack (a recursive function). Alternatively, the playmessage() function can return control to its caller (perhaps a while loop that spins forever) and pass back a code to indicate the command. The caller acts accordingly. This is non-recursive, but for anything but simple applications this style becomes tedious as you start needing to pass back more info and up longer chains of functions. Any guidance on this would be appreciated. Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/392ebb7f/attachment-0002.html From Prometheus001 at gmx.net Fri Dec 4 08:01:44 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 17:01:44 +0100 Subject: [Freeswitch-users] Voicmail - message only Message-ID: <4B193268.20009@gmx.net> Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter From nik.middleton at noblesolutions.co.uk Fri Dec 4 08:06:57 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 16:06:57 -0000 Subject: [Freeswitch-users] B Leg on bridged call is not hanging up Message-ID: Hi Guys, This one has me stumped. I'm originating a call, playing audio, trapping on DTMF and bridging to another endpoint (read phone number) If the A leg hangs up, then the call is cleared down and all is well. However if the B Leg attempts to hang-up, the LUA script that is handling the bridge continues to play audio to the a leg, while the B leg is in limbo. It does eventually time out with no RTP. Running Sofia debug on the cli shows that I'm getting the BYE from the B Leg, but that's about as far as I can get. The hang-up hook is not being fired in the lua script. Anyone give me some pointers as to where I might start looking? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/06930b44/attachment-0002.html From frank at carmickle.com Fri Dec 4 08:47:46 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 4 Dec 2009 11:47:46 -0500 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B193268.20009@gmx.net> References: <4B193268.20009@gmx.net> Message-ID: <20091204164746.GA31924@base.carmickle.com> Hello On Fri, Dec 04, Peter P GMX wrote: > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? Do you mean from the wiki http://wiki.freeswitch.org/wiki/Mod_voicemail#skip_instructions skip_instructions Skips playback of instructions when leaving messages. Variable is unset after voicemail application finishes. --FC From lists at redbonez.net Fri Dec 4 08:50:00 2009 From: lists at redbonez.net (Adam Ford) Date: Fri, 4 Dec 2009 09:50:00 -0700 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B193268.20009@gmx.net> References: <4B193268.20009@gmx.net> Message-ID: <01e801ca7501$d2b49eb0$781ddc10$@net> I am still new to freeswitch, but I would think you could achieve this by just passing the call to an IVR application that plays the message instead of passing it to the voicemail application. -AF -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Friday, December 04, 2009 9:02 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Voicmail - message only Hello, is there a chance to have the voicemail system to play announcment #1 only and not play announcement and then record the voicemail? Means: Can I switch off the recording part? Best regards Peter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From frank at carmickle.com Fri Dec 4 09:19:27 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 4 Dec 2009 12:19:27 -0500 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <01e801ca7501$d2b49eb0$781ddc10$@net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> Message-ID: <20091204171927.GB31924@base.carmickle.com> On Fri, Dec 04, Adam Ford wrote: > I am still new to freeswitch, but I would think you could achieve this by > just passing the call to an IVR application that plays the message instead > of passing it to the voicemail application. > > -AF > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P > GMX > Sent: Friday, December 04, 2009 9:02 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Voicmail - message only > > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? Yeah. I guess it was unclear to me which part he wanted to switch off. You could just use playback or play_and_get_digits. --FC From Prometheus001 at gmx.net Fri Dec 4 09:26:14 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 18:26:14 +0100 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <01e801ca7501$d2b49eb0$781ddc10$@net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> Message-ID: <4B194636.7030306@gmx.net> I would like to manage this in the voicemail menu. "Press 6 to enable recording" "Press 7 to only play announcement" or so. So hte user can manage it's settings on his own. Best regrds Peter Adam Ford schrieb: > I am still new to freeswitch, but I would think you could achieve this by > just passing the call to an IVR application that plays the message instead > of passing it to the voicemail application. > > -AF > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter P > GMX > Sent: Friday, December 04, 2009 9:02 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Voicmail - message only > > Hello, > > is there a chance to have the voicemail system to play announcment #1 > only and not play announcement and then record the voicemail? > Means: Can I switch off the recording part? > > Best regards > Peter > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Fri Dec 4 09:33:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Dec 2009 09:33:32 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting! Message-ID: <87f2f3b90912040933h568df38ch87ca88c205d88e8f@mail.gmail.com> FYI, The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04 Please call in! :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/d981579b/attachment-0002.html From lfurrea at gmail.com Fri Dec 4 10:16:14 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 4 Dec 2009 12:16:14 -0600 Subject: [Freeswitch-users] Sporadic call drops Message-ID: Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200 at 172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200 at 172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 200 at 172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/200 at 172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/5a512b1b/attachment-0002.html From freeswitch at cartissolutions.com Fri Dec 4 11:32:56 2009 From: freeswitch at cartissolutions.com (Yossi Neiman) Date: Fri, 04 Dec 2009 13:32:56 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Message-ID: <4B1963E8.7050204@cartissolutions.com> A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as to troubleshoot a problem in FreeSWITCH. It's extremely frustrating to try to help people out who won't try the provided suggestions first. And note directly to "eaf" - bogomips is quite possibly the least significant bit of data about a cpu that you will get out of /proc/cpuinfo... The name itself - bogo, means bogus. http://en.wikipedia.org/wiki/Bogomips -Yossi From anthony.minessale at gmail.com Fri Dec 4 11:48:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 13:48:08 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <4B1963E8.7050204@cartissolutions.com> References: <26594250.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> Message-ID: <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> There is another user here with a 300mhz box. I am willing to investigate this improved performance for weak devices but I need to do it in a sane cross-platform way. On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman wrote: > A word to the wise to the general FreeSWITCH community: If Anthony > Minessale suggests that you try to do any number of things, it's a very > good idea to try all those ideas before continuing on. I've known him, > MikeJ, and bkw for several years, and they almost always have very good > ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > frustrating to try to help people out who won't try the provided > suggestions first. > > And note directly to "eaf" - bogomips is quite possibly the least > significant bit of data about a cpu that you will get out of > /proc/cpuinfo... The name itself - bogo, means bogus. > http://en.wikipedia.org/wiki/Bogomips > > -Yossi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/1b13c721/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 11:51:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 13:51:48 -0600 Subject: [Freeswitch-users] Sporadic call drops In-Reply-To: References: Message-ID: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> we changed that message a long time ago so people would not think that anymore We are now 3000 rev beyond the version you are at, I would like it if you try the lastest trunk. On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea wrote: > Hi all, > > Guys I know the question could be too vague, but I have a customer that > just reported frequent failure to place outbound calls though a PSTN gateway > on the LAN. > > I looked at the logs and I seem to be able to confirm that FS fails to > place the call through the gateway and that the issue resides on the FS side > since the first channel that s killed is tht of the internal extension > registered to FS and then FS send the BYE to gw and kills the channel. > > What are possible causes of this? > > I know you always like to look at complete logs but here's a snip that > could shed some light on the disconnection. (I can provide full logs if > required and worthed) > > 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/internal/200 at 172.16.3.5 entering state [ready][200] > 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel > sofia/internal/200 at 172.16.3.5 entering state [terminated][200] > 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup > sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] > 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal sofia/internal/ > 200 at 172.16.3.5 [BREAK] > 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() > sofia/internal/200 at 172.16.3.5 ending bridge by request from write function > 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() > sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] > > > Is the 6th line normal behavior for ending the channel? > > FreeSWITCH Version 1.0.trunk (13484M) > > TIA > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/bf8bf8fe/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 12:00:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:00:25 -0600 Subject: [Freeswitch-users] B Leg on bridged call is not hanging up In-Reply-To: References: Message-ID: <191c3a030912041200k12c46c8dufe6573eac25bba43@mail.gmail.com> did you set the channel variable hangup_after_bridge=true on the A leg? On Fri, Dec 4, 2009 at 10:06 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > This one has me stumped. > > I'm originating a call, playing audio, trapping on DTMF and bridging to > another endpoint (read phone number) > > If the A leg hangs up, then the call is cleared down and all is well. > However if the B Leg attempts to hang-up, the LUA script that is handling > the bridge continues to play audio to the a leg, while the B leg is in > limbo. It does eventually time out with no RTP. > > Running Sofia debug on the cli shows that I'm getting the BYE from the B > Leg, but that's about as far as I can get. The hang-up hook is not being > fired in the lua script. > > Anyone give me some pointers as to where I might start looking? > > regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/23929090/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 12:01:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:01:50 -0600 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <4B194636.7030306@gmx.net> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> <4B194636.7030306@gmx.net> Message-ID: <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> You could file it as a feature request and post a bounty and probably get the functionality fairly inexpensively maybe $100 On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX wrote: > I would like to manage this in the voicemail menu. > "Press 6 to enable recording" > "Press 7 to only play announcement" > or so. So hte user can manage it's settings on his own. > > Best regrds > Peter > > Adam Ford schrieb: > > I am still new to freeswitch, but I would think you could achieve this by > > just passing the call to an IVR application that plays the message > instead > > of passing it to the voicemail application. > > > > -AF > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Peter P > > GMX > > Sent: Friday, December 04, 2009 9:02 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Voicmail - message only > > > > Hello, > > > > is there a chance to have the voicemail system to play announcment #1 > > only and not play announcement and then record the voicemail? > > Means: Can I switch off the recording part? > > > > Best regards > > Peter > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/66a55479/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Dec 4 12:03:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 20:03:46 -0000 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge Message-ID: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/98eb7cbe/attachment-0002.html From jerry.richards at teotech.com Fri Dec 4 12:47:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 4 Dec 2009 12:47:41 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Message-ID: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry From anthony.minessale at gmail.com Fri Dec 4 12:56:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:56:33 -0600 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge In-Reply-To: References: Message-ID: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi, > > > > Is there an option to hang-up both call legs in a bridge when one leg hangs > up? > > > > In my lua script I only ever see the hang-up for the call I?m in, not for > the bridged b leg. That said, I can see both a hang-up and un bridge event > being fired for the B leg. However my issue is that the A leg is still up, > and if I?ve called 2 Pots numbers, the phone network will maintain the > bridge. > > > > Is my only option to subscribe to the unbridge event and fire a hang-up > event using the ?other leg? UID? > > > > Regards, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/b3a7c9e0/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 12:59:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 14:59:38 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> Message-ID: <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: > > I have Mediant 1000 gateway, and for some reason, when I make an outbound > call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A > Wireshark trace shows that FS is replying to the gateway's inbound RTP > packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP > packets to the same port that FS specified in the outbound INVITE. It > appears in the log that FS is discarding the 200 OK from the gateway. > > I disabled the Firewall and SELinux on the Freeswitch machine. I tried > changing to "true" and also "proxy", but it has no effect. > > Anyone know what could be the issue? I posted the Freeswitch log in the > pastebin. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/c1c71f9f/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Dec 4 13:16:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 4 Dec 2009 21:16:58 -0000 Subject: [Freeswitch-users] Option to hang-up both legs in a bridge In-Reply-To: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> References: <191c3a030912041256t77dcee17t7ae0d5cca1ef09af@mail.gmail.com> Message-ID: Thanks for that, no didn't see the message, there seems to be a big delay in the messages getting turned around on the list. Yup, works great thanks. Script doesn't get events, so there was no way to check for the b leg hang-up. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 04 December 2009 20:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Option to hang-up both legs in a bridge did you see my reply to the other thread? set the channel variable hangup_after_bridge=true on the a leg your script must not be checking for the case when b leg hangs up that A leg does not hangup unless that var is set. On Fri, Dec 4, 2009 at 2:03 PM, Nik Middleton wrote: Hi, Is there an option to hang-up both call legs in a bridge when one leg hangs up? In my lua script I only ever see the hang-up for the call I'm in, not for the bridged b leg. That said, I can see both a hang-up and un bridge event being fired for the B leg. However my issue is that the A leg is still up, and if I've called 2 Pots numbers, the phone network will maintain the bridge. Is my only option to subscribe to the unbridge event and fire a hang-up event using the 'other leg' UID? Regards, _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/db24fb60/attachment-0002.html From pjintheusa at gmail.com Fri Dec 4 13:29:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 4 Dec 2009 16:29:27 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system Message-ID: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/7d72dd95/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 14:16:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 16:16:00 -0600 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Message-ID: <191c3a030912041416t1345736fs5548e7ca889006bb@mail.gmail.com> you could make an endpoint module for FS that speaks the special protocol then use that to call the conference. On Fri, Dec 4, 2009 at 3:29 PM, Phillip Jones wrote: > Hi All, > > Every so often you have to ask a question - where you know so little - it's > hard to even now where to start. This is one of the times. I am not > expecting an full answer here, just a gentle nudge in right direction to get > me started. > > What I have is a propriety IP based conference system - who want to add the > ability to have inbound PSTN callers join their conferences. All their > signaling is propriety - no SIP - but I do have access to that signaling > schema so can do some translation. Enough to get the IP / Port & CODEC of > the RTP stream. They use speex rtp sessions over TCP. > > So from an architectural point of view I am thinking of having the callers > enter a FS conference and than bridge that conference to their IP based > conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a Sofia > gateway) to that IP based conference system I am kind of lost. But it seems > reasonable that I should be able to get my head round this, because I know > the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/5410219c/attachment-0002.html From mgg at giagnocavo.net Fri Dec 4 14:16:50 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 4 Dec 2009 17:16:50 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/47ecb171/attachment-0002.html From kristian.kielhofner at gmail.com Fri Dec 4 14:41:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 4 Dec 2009 17:41:12 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> Message-ID: <2d9149cd0912041441v5b5ef62bjf9bd5afe833f20a5@mail.gmail.com> A little more data from one of my (our) boxes: starbox_352 ~ # uname -a Linux starbox_352 2.6.26.8-astlinux #1 PREEMPT Tue Nov 24 16:20:52 EST 2009 i586 unknown starbox_352 ~ # starbox_352 ~ # cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 5 model : 10 model name : Geode(TM) Integrated Processor by AMD PCS stepping : 2 cpu MHz : 498.053 cache size : 128 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de pse tsc msr cx8 sep pge cmov clflush mmx mmxext 3dnowext 3dnow bogomips : 997.21 clflush size : 32 power management: starbox_352 ~ # cat /etc/astlinux-release astlinux-s2s-3491 starbox_352 ~ # I'll find one that has been in production for a while with some active calls... On Thu, Dec 3, 2009 at 6:49 PM, Anthony Minessale wrote: > Sigh, > > You just took it up a notch in terms of disdain and sarcasm. > Why do people always only apologize sarcastically? > > I asked you to try the -hp and turn off the monotonic clock just to gather > the results to help you.? You completely missed it and just went on about > the threads.?? Please save the "ok fine the code is perfect, blah blah" if > you would have just read the email and answered the question I might have > cared more about the status of your problem. > > I told you both of those threads need to be on their toes because they try > to balance between a certian number of sql stmts or 500ms whatever comes > first.? When there are thousands of events per second being turned into SQL > statements which are in turn compiled into large sql transactions. > > If you want to come up with a way that they can sleep longer until there is > a sign of activity and stay busy for a few seconds then slow down again, > that's probably possible but the process is already idle at 0% cpu so maybe > you can appreciate why we are not rushing to work on it.? Maybe I'll give it > a go just to show you it has nothing to do with your problem. > > Please don't mock our comment about several years.? You have no idea how > hard this code was to develop and it's truly insulting.? Its clear to see > you are locked into assuming that the busy threads that are not all that > busy because they are constantly yielding to the scheduler is breaking the > timing code.? I begged you to understand me when i told you that the err is > not normal, most boxes do not see it doing nothing and there has to be a > specific problem on your box or configuration.? So instead of working with > us you want to escalate to snotty comments.? That's pretty normal on the > internet I guess.....? If you want to have a constructive conversation about > our core, install FS on a normal box, use it for a few weeks, figure out > everything about how it works then try.... There was pure speculation and > conjecture in your original emails and I never said a word about it until > you kept pushing. > > Kristian mentioned he never sees that on that same hardware did you even > consider following up on why that is? > > I don't have your device, but I assume if you get it working well it will > certainly help you more than it helps me so you could at least have the > decency to believe what we are trying to tell you. > > > > > > > > On Thu, Dec 3, 2009 at 3:44 PM, eaf wrote: >> >> Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do >> that. >> At the moment, I hope it won't be necessary as I can make those "hyper" >> threads behave, and will see how that goes first. I see where your >> implementation could be coming from. There is a queue of SQL queries in >> sofia.c processed by the worker thread. There are only two pop functions >> available in APR: queue_pop() and queue_trypop(), so alas no option with a >> timeout here. You don't want to block the thread in pop() indefinitely >> because you chose that same worker needs to do ireg and gw processing once >> in a while (separated by tens or hundreds of seconds, btw). You also want >> to >> be able to detect shutdown condition so that the worker doesn't hold up >> profile thread. So you chose to poll for events every millisecond instead >> of >> just creating an apr_thread_cond_t for resource friendly signalling. >> >> I agree that the timer thread philosophy is great and was the right choice >> for scaling, but I just don't comprehend responses to things like these >> other SQL or sofia worker threads. Did somebody even remotely acknowledge >> that busy loops at least in those areas that I showed may probably be a >> bad >> idea and could've been eliminated? I've heard suggestions to bump up >> priority, I've heard that the code was perfect already, that it's the >> result >> of 4-year effort, that I am arrogant, don't listen and don't understand >> squat. >> >> I'm sorry if I gave you impression that I was looking for the bad parts in >> the software. I apologized for that already. All I wanted was to have >> constructive conversation, perhaps I'm not too good at it. Code is already >> perfect according to you? Fine with me. >> >> >> Anthony Minessale-2 wrote: >> > >> > no, >> > >> > I mean the one after that that you must have completely skipped with a >> > command line option to try and a param to set in the config. It somewhat >> > annoys me for taking the time to compose it now. ?I wrote all of the >> > code >> > you are talking about myself and I was trying to give you some >> > suggestions.... >> > >> > Well, actually, ?you did answer my question about the platform so you >> > must >> > have seen it..... >> > >> > The loops are not the cause of that migration message, something wrong >> > with >> > the hardware or the kernel is. >> > Another guy just told you he does not see that problem on the same exact >> > hardware. >> > >> > Even if you have a point about the sql threads, you could make a patch >> > to >> > slow them down but you cant slow down too much or you will not be able >> > to >> > handle 400 cps all asking to send updates to transactions in batches of >> > thousands of sql stmts. ?Every line of that code is carefully designed >> > so >> > I >> > don't know what else to tell you but to stop being so arrogant and >> > re-read >> > this thread for all the advice you have totally ignored. ?I started out >> > trying to help you but I have a lot of work to do. ?I thoroughly >> > explained >> > it to you and you are choosing to ignore me so I guess I'm done. >> > You can do whatever you want with your working copy, i'll see you in 3 >> > or >> > 4 >> > years when you get up to speed with the rest of us........ >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26633739.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From mayamatakeshi at gmail.com Fri Dec 4 14:45:17 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 5 Dec 2009 07:45:17 +0900 Subject: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS In-Reply-To: References: Message-ID: <15b9404e0912041445m19c5640avfc43bd17c960ea68@mail.gmail.com> I had this same problem today. I solved it using OPTION = 67108864 instead of OPTIONS = 67108864 I'm using CentOS5.3 (x86_64) br, takeshi On Sat, Nov 28, 2009 at 12:36 AM, Frank @ Impact wrote: > Yes. I am using version 5.1 I am using Fedora 12. > > > > -----Original Message----- > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Leon de > Rooij > *Sent:* Friday, November 27, 2009 10:19 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > > > Are you using the myodbc 3.51.18 version or higher ? > > > > I'm using 3.51.19 (ubuntu karmic) and it works properly. I also had to > upgrade from jaunty.. > > > > regards, > > > > Leon > > > > > > On Nov 27, 2009, at 3:41 PM, Frank @ Impact wrote: > > > > Thanks. But when I made these entries in /etc/odbc.ini and rebooted? > > > > [freeswitch] > > Driver = MySQL > > SERVER = 127.0.0.1 > > PORT = 4040 > > DATABASE = mydb > > OPTIONS = 67108864 > > > > ?I still get FS complaining with this. > > > > Nov 27 08:45:57 P3 freeswitch[27933]: 2009-11-27 08:45:57.016744 [WARNING] > sofia_glue.c:3918 GREAT SCOTT!!! Cannot execute batched statements!#012If > you are using mysql, make sure you are using MYODBC 3.51.18 or higher and > enable FLAG_MULTI_STATEMENTS > > > > FreeSWITCH>version > > FreeSWITCH Version 1.0.trunk (15660) > > > > Linux P3.dom.com 2.6.30.9-96.fc11.x86_64 #1 SMP Wed Nov 4 00:02:04 EST > 2009 x86_64 x86_64 x86_64 GNU/Linux > > > > From /etc/odbcinst.ini > > DRIVER = /usr/lib64/libmyodbc5-5.1.5.so > > Setup = /usr/lib64/libodbcmyS.so > > > > Is this a FS issue ? or an issue with mysql odbc? Any insight would be > great. > > > > -----Original Message----- > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Leon de Rooij > *Sent:* Friday, November 27, 2009 3:37 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS > > > > There's a little info here on how to enable it with odbc: > > > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#CentOS_5.2 > > > > regards, > > > > Leon > > > > > > On Nov 26, 2009, at 10:48 PM, Tihomir Culjaga wrote: > > > > > > > On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris wrote: > > http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html > > > > MySQL Connector/ODBC now supports batched statements. In order to enable > > cached statement support you must switch enable the batched > > statement option (FLAG_MULTI_STATEMENTS, > > 67108864, or Allow multiple statements > > within a GUI configuration). Be aware that batched statements > > create an increased chance of SQL injection attacks and you must > > ensure that your application protects against this scenario. > > (Bug#7445 ) > > > > > so, is this the right patch ? > > http://bugs.mysql.com/file.php?id=6994 > > > T. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/0fe3022d/attachment-0002.html From Prometheus001 at gmx.net Fri Dec 4 14:51:31 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Dec 2009 23:51:31 +0100 Subject: [Freeswitch-users] Voicmail - message only In-Reply-To: <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> References: <4B193268.20009@gmx.net> <01e801ca7501$d2b49eb0$781ddc10$@net> <4B194636.7030306@gmx.net> <191c3a030912041201l6f6c6313n532522a48d6418aa@mail.gmail.com> Message-ID: <4B199273.6090301@gmx.net> Hello Anthony, thanks for the hint. I have posted a $100 bounty in the wiki + another $150 bounty to enable speaking an announcement via TTS. Best regards Peter Anthony Minessale schrieb: > You could file it as a feature request and post a bounty and probably > get the functionality fairly inexpensively maybe $100 > > > > On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX > wrote: > > I would like to manage this in the voicemail menu. > "Press 6 to enable recording" > "Press 7 to only play announcement" > or so. So hte user can manage it's settings on his own. > > Best regrds > Peter > > Adam Ford schrieb: > > I am still new to freeswitch, but I would think you could > achieve this by > > just passing the call to an IVR application that plays the > message instead > > of passing it to the voicemail application. > > > > -AF > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf > Of Peter P > > GMX > > Sent: Friday, December 04, 2009 9:02 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: [Freeswitch-users] Voicmail - message only > > > > Hello, > > > > is there a chance to have the voicemail system to play > announcment #1 > > only and not play announcement and then record the voicemail? > > Means: Can I switch off the recording part? > > > > Best regards > > Peter > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pjintheusa at gmail.com Fri Dec 4 14:58:52 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 4 Dec 2009 17:58:52 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> Message-ID: <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo wrote: > I think you will need to sort out the signaling first, as you?ll have to > tell the conference system to accept which RTP streams for which > conferences, as well as tell it to transmit to your callers, no? > > > > After that, then I would imagine you just need to do SDP rewriting when a > call hits FreeSWITCH. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Friday, December 04, 2009 2:29 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Bridging to a non SIP based system > > > > Hi All, > > Every so often you have to ask a question - where you know so little - it's > hard to even now where to start. This is one of the times. I am not > expecting an full answer here, just a gentle nudge in right direction to get > me started. > > What I have is a propriety IP based conference system - who want to add the > ability to have inbound PSTN callers join their conferences. All their > signaling is propriety - no SIP - but I do have access to that signaling > schema so can do some translation. Enough to get the IP / Port & CODEC of > the RTP stream. They use speex rtp sessions over TCP. > > So from an architectural point of view I am thinking of having the callers > enter a FS conference and than bridge that conference to their IP based > conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a Sofia > gateway) to that IP based conference system I am kind of lost. But it seems > reasonable that I should be able to get my head round this, because I know > the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/133343b9/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 15:41:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 17:41:19 -0600 Subject: [Freeswitch-users] Playing an rtp stream In-Reply-To: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> References: <367751820912031747j31841b07wb3bab8a11920ec36@mail.gmail.com> Message-ID: <191c3a030912041541j12732ef6t1577cf550c811375@mail.gmail.com> yes this is possible assuming that is a either a multicast address or a dedicated unicast address you want to listen on that something else is sending audio to. it would also require writing a module in C to actually implement it. On Thu, Dec 3, 2009 at 7:47 PM, Phillip Jones wrote: > Hi there, > > It it possible do something like: > > > > > > > > > > Basically I have need to connect to incoming calls listen to an existing > rtp stream - I know the IP and port. > > Any hints on achieving this would be much appreciated. > > Thanks > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/3076a59b/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 4 15:48:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Dec 2009 17:48:21 -0600 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: References: Message-ID: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> set rotate-on-hup to false in the cdr_csv config file then it will only rotate when the file gets too big and also you can get a cdr with session.generateXmlCdr() and dig out what you need or get it from variables but it will not be nearly as reliable as using the C ones because you need low level access to make sure you write to the disk properly from many threads etc. On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane wrote: > is it possible to run a javascript at the end of dialplan to generate cdrs? > because (mod_cdr_csv) is giving me hard time as it rotates Master file on > machine reboots or shutdown signals. > javascript or LUA for preferences? > > thank you > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/98c28996/attachment-0002.html From msc at freeswitch.org Fri Dec 4 16:35:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Dec 2009 16:35:45 -0800 Subject: [Freeswitch-users] errors installing wanpipe drivers In-Reply-To: References: Message-ID: <87f2f3b90912041635s134375d0ife1b5a76e337f097@mail.gmail.com> Looks good so far. Try "oz list" and "oz dump 1" and see what happens. -MC On Thu, Dec 3, 2009 at 10:36 PM, Neil Patel wrote: > Thanks all for your help. I got around this by running ./Setup and > installing wanpipe in TDM API mode (it says it's the default for FS). I then > uncommented the mod_openzap line in modules.conf when installing FS. Finally > I ran wancfg_fs which creates appropriate config files for you for your FS > installation. I believe openzap is now installed properly: > > 2009-12-04 12:04:52.411017 [INFO] zap_io.c:2451 Loading IO from > /usr/local/freeswitch/mod/ozmod_wanpipe.so [wanpipe] > 2009-12-04 12:04:52.411126 [INFO] zap_io.c:2251 auto-loaded 'wanpipe' > 2009-12-04 12:04:52.411311 [INFO] ozmod_wanpipe.c:287 configuring device > s1c1 as OpenZAP device 1:1 fd:14 DTMF: software > 2009-12-04 12:04:52.411377 [INFO] ozmod_wanpipe.c:287 configuring device > s1c2 as OpenZAP device 1:2 fd:15 DTMF: software > 2009-12-04 12:04:52.411444 [INFO] ozmod_wanpipe.c:287 configuring device > s1c3 as OpenZAP device 1:3 fd:17 DTMF: software > 2009-12-04 12:04:52.411509 [INFO] ozmod_wanpipe.c:287 configuring device > s1c4 as OpenZAP device 1:4 fd:18 DTMF: software > 2009-12-04 12:04:52.411575 [INFO] ozmod_wanpipe.c:287 configuring device > s1c5 as OpenZAP device 1:5 fd:19 DTMF: software > 2009-12-04 12:04:52.411639 [INFO] ozmod_wanpipe.c:287 configuring device > s1c6 as OpenZAP device 1:6 fd:20 DTMF: software > 2009-12-04 12:04:52.411707 [INFO] ozmod_wanpipe.c:287 configuring device > s1c7 as OpenZAP device 1:7 fd:21 DTMF: software > 2009-12-04 12:04:52.411771 [INFO] ozmod_wanpipe.c:287 configuring device > s1c8 as OpenZAP device 1:8 fd:22 DTMF: software > 2009-12-04 12:04:52.411837 [INFO] ozmod_wanpipe.c:287 configuring device > s1c9 as OpenZAP device 1:9 fd:23 DTMF: software > 2009-12-04 12:04:52.411903 [INFO] ozmod_wanpipe.c:287 configuring device > s1c10 as OpenZAP device 1:10 fd:24 DTMF: software > 2009-12-04 12:04:52.411969 [INFO] ozmod_wanpipe.c:287 configuring device > s1c11 as OpenZAP device 1:11 fd:25 DTMF: software > 2009-12-04 12:04:52.412034 [INFO] ozmod_wanpipe.c:287 configuring device > s1c12 as OpenZAP device 1:12 fd:26 DTMF: software > 2009-12-04 12:04:52.412102 [INFO] ozmod_wanpipe.c:287 configuring device > s1c13 as OpenZAP device 1:13 fd:27 DTMF: software > 2009-12-04 12:04:52.412179 [INFO] ozmod_wanpipe.c:287 configuring device > s1c14 as OpenZAP device 1:14 fd:28 DTMF: software > 2009-12-04 12:04:52.412244 [INFO] ozmod_wanpipe.c:287 configuring device > s1c15 as OpenZAP device 1:15 fd:29 DTMF: software > TDM API: CMD: 18 > : Operation not supported > 2009-12-04 12:04:52.412416 [INFO] ozmod_wanpipe.c:287 configuring device > s1c16 as OpenZAP device 1:16 fd:30 DTMF: none > 2009-12-04 12:04:52.412503 [INFO] ozmod_wanpipe.c:287 configuring device > s1c17 as OpenZAP device 1:17 fd:31 DTMF: software > 2009-12-04 12:04:52.412568 [INFO] ozmod_wanpipe.c:287 configuring device > s1c18 as OpenZAP device 1:18 fd:32 DTMF: software > 2009-12-04 12:04:52.412634 [INFO] ozmod_wanpipe.c:287 configuring device > s1c19 as OpenZAP device 1:19 fd:33 DTMF: software > 2009-12-04 12:04:52.412708 [INFO] ozmod_wanpipe.c:287 configuring device > s1c20 as OpenZAP device 1:20 fd:34 DTMF: software > 2009-12-04 12:04:52.412771 [INFO] ozmod_wanpipe.c:287 configuring device > s1c21 as OpenZAP device 1:21 fd:35 DTMF: software > 2009-12-04 12:04:52.412838 [INFO] ozmod_wanpipe.c:287 configuring device > s1c22 as OpenZAP device 1:22 fd:36 DTMF: software > 2009-12-04 12:04:52.412902 [INFO] ozmod_wanpipe.c:287 configuring device > s1c23 as OpenZAP device 1:23 fd:37 DTMF: software > 2009-12-04 12:04:52.412948 [INFO] ozmod_wanpipe.c:287 configuring device > s1c24 as OpenZAP device 1:24 fd:38 DTMF: software > 2009-12-04 12:04:52.412988 [INFO] ozmod_wanpipe.c:287 configuring device > s1c25 as OpenZAP device 1:25 fd:39 DTMF: software > 2009-12-04 12:04:52.413018 [INFO] ozmod_wanpipe.c:287 configuring device > s1c26 as OpenZAP device 1:26 fd:40 DTMF: software > 2009-12-04 12:04:52.413041 [INFO] ozmod_wanpipe.c:287 configuring device > s1c27 as OpenZAP device 1:27 fd:41 DTMF: software > 2009-12-04 12:04:52.413063 [INFO] ozmod_wanpipe.c:287 configuring device > s1c28 as OpenZAP device 1:28 fd:42 DTMF: software > 2009-12-04 12:04:52.413086 [INFO] ozmod_wanpipe.c:287 configuring device > s1c29 as OpenZAP device 1:29 fd:43 DTMF: software > 2009-12-04 12:04:52.413106 [INFO] ozmod_wanpipe.c:287 configuring device > s1c30 as OpenZAP device 1:30 fd:44 DTMF: software > 2009-12-04 12:04:52.413128 [INFO] ozmod_wanpipe.c:287 configuring device > s1c31 as OpenZAP device 1:31 fd:45 DTMF: software > 2009-12-04 12:04:52.413142 [INFO] zap_io.c:2374 Configured 31 channel(s) > 2009-12-04 12:04:52.431405 [INFO] zap_io.c:2468 Loading SIG from > /usr/local/freeswitch/mod/ozmod_ss7_boost.so > 2009-12-04 12:04:52.431441 [INFO] zap_io.c:2584 auto-loaded 'ss7_boost' > 2009-12-04 12:04:52.431541 [CONSOLE] switch_loadable_module.c:889 > Successfully Loaded [mod_openzap] > 2009-12-04 12:04:52.431553 [NOTICE] switch_loadable_module.c:142 Adding > Endpoint 'openzap' > 2009-12-04 12:04:52.431638 [NOTICE] switch_loadable_module.c:248 Adding > Application 'disable_ec' > 2009-12-04 12:04:52.431659 [NOTICE] switch_loadable_module.c:270 Adding API > Function 'oz' > 2009-12-04 12:04:52.432009 [WARNING] ss7_boost_client.c:244 TX EVENT (P): > SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0 > > > Does this look right? > Thanks. > > On Mon, Nov 30, 2009 at 9:09 PM, Moises Silva wrote: > >> On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel wrote: >> >>> Hi All, >>> >>> I am currently installing a Sangoma A102 card to work with FS using >>> wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get >>> openzap-related modules to compile: >>> >>> > cd wanpipe-3.5.6.5/ >>> > make openzap >>> ... >>> make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >>> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' >>> make -C api/libstelephony clean >>> make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >>> make[1]: *** No rule to make target `clean'. Stop. >>> make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' >>> make: *** [all_lib] Error 2 >>> >>> The libstelephony directory has no Makefile in it. Why is it missing? Is >>> there a version of wanpipe drivers that will work? I have been unsuccessful >>> with 3.4.4 and 3.5.6 in similar fashion. >>> >>> >> Hi Neil, >> >> Most likely the creation of the Makefile failed (since you mention you >> can't see a Makefile). Please be sure to have installed the pre-requisites >> listed at http://wiki.sangoma.com/Requirements >> >> Particularly in this case, libtool, autoconf and automake packages. >> >> -- >> Moises Silva >> Software Developer >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >> 9T3 Canada >> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/46fb4fb7/attachment-0002.html From mgg at giagnocavo.net Fri Dec 4 17:52:59 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 4 Dec 2009 20:52:59 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> Yes I was just thinking that it might be simpler to just fixup the SDP and just write some custom script to talk control to the backend conference system than to write a whole endpoint module. Especially cause you can do the fixup and control in a high level language (even if you use C#, you're going to end up playing with pointers except the syntax will be more verbose). Then again, I have a natural aversion to C so maybe it's just me ;) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 3:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Bridging to a non SIP based system Ah guys - that was exactly the nudge I was looking for - I will take a look at the other endpoint modules like mod_skypiax etc. I will also look at the SDP - I see where you are going there - I might not even need the conference in that case. Question is - could I write an endpoint is C# !!! :) Thanks again - that's a great help. On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo > wrote: I think you will need to sort out the signaling first, as you'll have to tell the conference system to accept which RTP streams for which conferences, as well as tell it to transmit to your callers, no? After that, then I would imagine you just need to do SDP rewriting when a call hits FreeSWITCH. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, December 04, 2009 2:29 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Bridging to a non SIP based system Hi All, Every so often you have to ask a question - where you know so little - it's hard to even now where to start. This is one of the times. I am not expecting an full answer here, just a gentle nudge in right direction to get me started. What I have is a propriety IP based conference system - who want to add the ability to have inbound PSTN callers join their conferences. All their signaling is propriety - no SIP - but I do have access to that signaling schema so can do some translation. Enough to get the IP / Port & CODEC of the RTP stream. They use speex rtp sessions over TCP. So from an architectural point of view I am thinking of having the callers enter a FS conference and than bridge that conference to their IP based conference room. That would do it. The problem is that because I can not bridge using SIP (through a Sofia gateway) to that IP based conference system I am kind of lost. But it seems reasonable that I should be able to get my head round this, because I know the IP / Port & CODEC of the RTP stream. But perhaps I missing a key bit of knowledge/understanding here. I would be grateful for any advise here. Thanks a lot, Phil _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/f153580f/attachment-0002.html From mrene_lists at avgs.ca Fri Dec 4 18:02:21 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 4 Dec 2009 21:02:21 -0500 Subject: [Freeswitch-users] Bridging to a non SIP based system In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> References: <367751820912041329w74208003hedcd0125a658bc77@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F776F@mse17be1.mse17.exchange.ms> <367751820912041458w4950350fq6114f4589fa2df17@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67032C1F77B4@mse17be1.mse17.exchange.ms> Message-ID: <3943EA1B-27CA-4400-96C4-CB6FD344B916@avgs.ca> You can re-use some of mod_sofia's functions (like sofia_glue_parse_sdp) and only write the part of signalling thats different from SIP. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 4-Dec-09, at 8:52 PM, Michael Giagnocavo wrote: > Yes I was just thinking that it might be simpler to just fixup the > SDP and just write some custom script to talk control to the backend > conference system than to write a whole endpoint module. Especially > cause you can do the fixup and control in a high level language > (even if you use C#, you?re going to end up playing with pointers > except the syntax will be more verbose). Then again, I have a > natural aversion to C so maybe it?s just me ;) > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Friday, December 04, 2009 3:59 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Bridging to a non SIP based system > > Ah guys - that was exactly the nudge I was looking for - I will take > a look at the other endpoint modules like mod_skypiax etc. I will > also look at the SDP - I see where you are going there - I might not > even need the conference in that case. > > Question is - could I write an endpoint is C# !!! :) > > Thanks again - that's a great help. > > On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo > wrote: > I think you will need to sort out the signaling first, as you?ll > have to tell the conference system to accept which RTP streams for > which conferences, as well as tell it to transmit to your callers, no? > > After that, then I would imagine you just need to do SDP rewriting > when a call hits FreeSWITCH. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Friday, December 04, 2009 2:29 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Bridging to a non SIP based system > > Hi All, > > Every so often you have to ask a question - where you know so little > - it's hard to even now where to start. This is one of the times. I > am not expecting an full answer here, just a gentle nudge in right > direction to get me started. > > What I have is a propriety IP based conference system - who want to > add the ability to have inbound PSTN callers join their conferences. > All their signaling is propriety - no SIP - but I do have access to > that signaling schema so can do some translation. Enough to get the > IP / Port & CODEC of the RTP stream. They use speex rtp sessions > over TCP. > > So from an architectural point of view I am thinking of having the > callers enter a FS conference and than bridge that conference to > their IP based conference room. That would do it. > > The problem is that because I can not bridge using SIP (through a > Sofia gateway) to that IP based conference system I am kind of lost. > But it seems reasonable that I should be able to get my head round > this, because I know the IP / Port & CODEC of the RTP stream. > > But perhaps I missing a key bit of knowledge/understanding here. > > I would be grateful for any advise here. > > Thanks a lot, > > > Phil > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/b98b7ef3/attachment-0002.html From andrewkt at aktzero.com Fri Dec 4 18:40:12 2009 From: andrewkt at aktzero.com (Andrew Thompson) Date: Fri, 04 Dec 2009 21:40:12 -0500 Subject: [Freeswitch-users] Eavesdrop error? In-Reply-To: <03c401ca73bf$1cea8600$56bf9200$@com> References: <036401ca739b$dc451340$94cf39c0$@com> <03b101ca73a6$4c8db080$e5a91180$@com> <191c3a030912021534m588ddcat2539c20c42ee537d@mail.gmail.com> <03c401ca73bf$1cea8600$56bf9200$@com> Message-ID: <4B19C80C.2060508@aktzero.com> On 12/2/2009 9:19 PM, Lars Zeb wrote: > Is this reasonable given it was the only call in FreeSwitch at the time? How > can this situation be corrected in the future? As a workaround, you can eavesdrop with 779, and use * to navigate channels. -- Andrew Thompson From pmhshz at gmail.com Fri Dec 4 23:17:36 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 4 Dec 2009 23:17:36 -0800 (PST) Subject: [Freeswitch-users] Need Conference design help Message-ID: <26653473.post@talk.nabble.com> Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3. User who can only participate when Moderator allow them to get in. Also besides above setup I have to perform other things like Record the conference, Multicast the conference to other freeswitch server. I saw the conference Record CLI command but wondering where to setup when conference starts. I am also wondering how Multicast Conference is possible in Freeswitch and how the receiver Freeswitch configuration will look like. Thanks. msp -- View this message in context: http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rsavage at KingBallow.com Fri Dec 4 12:07:14 2009 From: rsavage at KingBallow.com (Reece Savage) Date: Fri, 4 Dec 2009 14:07:14 -0600 Subject: [Freeswitch-users] Aastra XML scripts. Message-ID: <8E4ACA7747F7F641991455BC157390C80145007D@srv-nash-ex.mail.kingballow.com> Would anyone be willing to port the Aastra XML scripts for Asterisk to FreeSWITCH? I would be willing to sponser. Reece Savage Information Technology Manager King & Ballow Law Offices 315 Union Street Suite 1100 Nashville, TN 37201 Phone (615) 726-5525 Fax (615) 254-7907 rsavage at kingballow.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091204/fcfad003/attachment-0002.html From mailinglist at fribert.dk Sat Dec 5 01:46:30 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sat, 05 Dec 2009 10:46:30 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? Message-ID: <4B1A3A06020000E100000289@mail.fribert.dk> Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/c260b101/attachment-0002.html From testeador01 at gmail.com Sat Dec 5 07:02:00 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 5 Dec 2009 10:02:00 -0500 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: Hello and welcome to FreeSWITCH, This is the starter's guide: http://wiki.freeswitch.org/wiki/Getting_Started_Guide Also Michael Collins wrote this nice article that will help you get started in VoIP and Freeswitch: http://bit.ly/EpVrv Most of the FreeSWITCH features are documented in the wiki, although I suggest not searching in the wiki search box but using google. 2009/12/5 mailinglist > Has anybody done this? > > I'm completely at a loss, having tinkered very little with Asterisk, and > giving up on that, I wonder if there's any help to be found on FreeSwitch? > Anybody that can give pointers to a good step-by-step instruction? > > I want to have it handle my two sip-phones (siemens dect ip and spa 901), > and handle a sip account at my provider. > Of course transferring calls between the two, as well as group calls would > be a nice benefit. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/5130f644/attachment-0002.html From talk2ram at gmail.com Sat Dec 5 07:22:56 2009 From: talk2ram at gmail.com (ram) Date: Sat, 5 Dec 2009 20:52:56 +0530 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: On Sat, Dec 5, 2009 at 3:16 PM, mailinglist wrote: > Has anybody done this? > > I'm completely at a loss, having tinkered very little with Asterisk, and > giving up on that, I wonder if there's any help to be found on FreeSwitch? > Anybody that can give pointers to a good step-by-step instruction? > > I want to have it handle my two sip-phones (siemens dect ip and spa 901), > and handle a sip account at my provider. > Of course transferring calls between the two, as well as group calls would > be a nice benefit. > > > in short answer Fusionpbx.com Ram > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/67b628b5/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Dec 5 07:41:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 15:41:34 -0000 Subject: [Freeswitch-users] how to disable hook flash hold Message-ID: Hi, Is it possible to disable being able to put a call on hold using hook flash? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/b14a698f/attachment-0002.html From testeador01 at gmail.com Sat Dec 5 08:05:02 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 5 Dec 2009 11:05:02 -0500 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi,? Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pjintheusa at gmail.com Sat Dec 5 08:49:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Sat, 5 Dec 2009 11:49:27 -0500 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall??? In-Reply-To: References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: <367751820912050849s2e0acd45we4c60989af78f8bd@mail.gmail.com> Also check out this great write up: http://wiki.freeswitch.org/wiki/Multi_home_tutorial This is pfSense specific. On Sat, Dec 5, 2009 at 10:22 AM, ram wrote: > > > On Sat, Dec 5, 2009 at 3:16 PM, mailinglist wrote: > >> Has anybody done this? >> >> I'm completely at a loss, having tinkered very little with Asterisk, and >> giving up on that, I wonder if there's any help to be found on FreeSwitch? >> Anybody that can give pointers to a good step-by-step instruction? >> >> I want to have it handle my two sip-phones (siemens dect ip and spa 901), >> and handle a sip account at my provider. >> Of course transferring calls between the two, as well as group calls would >> be a nice benefit. >> >> >> > > in short answer Fusionpbx.com > > Ram > > > >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/c5c8c9f3/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Dec 5 09:20:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 17:20:33 -0000 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: Sorry, I meant from a POTS phone Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: 05 December 2009 16:05 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi,? Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lists at redbonez.net Sat Dec 5 09:52:37 2009 From: lists at redbonez.net (Adam Ford) Date: Sat, 5 Dec 2009 10:52:37 -0700 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1A3A06020000E100000289@mail.fribert.dk> References: <4B1A3A06020000E100000289@mail.fribert.dk> Message-ID: <0E0013F55E224674A1361329CF7A85F0@redbonez> I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office. http://doc.pfsense.org/index.php/FreeSWITCH -AF _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mailinglist Sent: Saturday, December 05, 2009 2:47 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/80f712b1/attachment-0002.html From tculjaga at gmail.com Sat Dec 5 11:01:10 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 5 Dec 2009 20:01:10 +0100 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: References: Message-ID: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> The POTS phone is attached to something... (ZAP channel or an ATA or a gateway). It is there you configure this behaviour. T. On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Sorry, I meant from a POTS phone > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena > Sent: 05 December 2009 16:05 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] how to disable hook flash hold > > It can be done from the phone itself; for example on a Grandstream > phone it is done with the option "Onhook Threshold:" setting it to > "hookflash OFF" > > > 2009/12/5 Nik Middleton > > > > Hi, Is it possible to disable being able to put a call on hold using > hook flash? > > > > > > > > Regards > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/ecded617/attachment-0002.html From pekkis50 at gmail.com Sat Dec 5 11:06:19 2009 From: pekkis50 at gmail.com (Pekka Kurki) Date: Sat, 05 Dec 2009 20:06:19 +0100 Subject: [Freeswitch-users] freeswitch binaries w/o IPv6 anywhere (for w2k)? Message-ID: <4B1AAF2B.1070305@gmail.com> thanks, --pekka-- From nik.middleton at noblesolutions.co.uk Sat Dec 5 11:32:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 5 Dec 2009 19:32:51 -0000 Subject: [Freeswitch-users] how to disable hook flash hold In-Reply-To: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> References: <65d96fc80912051101v2958b273qc14edd1e1cfedf6e@mail.gmail.com> Message-ID: It's a pots phone at the end of a VoIP trunk provided by my ISP. I have not control over it. The only think I have found so far is: Which is what I presume I add to my provider's conf file. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: 05 December 2009 19:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold The POTS phone is attached to something... (ZAP channel or an ATA or a gateway). It is there you configure this behaviour. T. On Sat, Dec 5, 2009 at 6:20 PM, Nik Middleton wrote: Sorry, I meant from a POTS phone Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Milena Sent: 05 December 2009 16:05 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] how to disable hook flash hold It can be done from the phone itself; for example on a Grandstream phone it is done with the option "Onhook Threshold:" setting it to "hookflash OFF" 2009/12/5 Nik Middleton > > Hi, Is it possible to disable being able to put a call on hold using hook flash? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/af67cd72/attachment-0002.html From Prometheus001 at gmx.net Sat Dec 5 12:35:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 05 Dec 2009 21:35:28 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> Message-ID: <4B1AC410.9050201@gmx.net> Hello Anthony, I did some checks today Here is how the phones are registered: mysql> select sip_host, presence_hosts, server_user,server_host, hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip_host | presence_hosts | server_user | server_host | hostname | sip_realm | mwi_user | mwi_host | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ IPs are: 10.11.12.1 sip1.mydomain.com (common cluster IP) 10.11.12.2 sip11.mydomain.com 10.11.12.3 sip12.mydomain.com (not used at this point) XML-Curl for the directory is:
The internal profile has the following alias: With $${domain} being sip11.mydomain.com Phones are registering to sip1.mydomain.com, Voicemail works, but MWI does not. Any hint what I should change to make this work? Best regards Peter Anthony Minessale schrieb: > based on your example past > > sip1.mydomain.com is the domain in the > packet and thus the profile should have an alias for this. > Then the user must reside in your sip db with the user 200 and domain > sip1.mydomain.com > > if you dont have this consider the force-register-domain and > force-register-db-domain to normalize the host names. > > > On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale > > wrote: > > Did you check the 2 replies that told you you need aliases in your > sofia profile to translate the domain found in your > message_waiting to the right profile? Both Brian and Mike > answered you. > > > > > > On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX > > wrote: > > I tried now with phones directly attached to the freeswitch > (without an > OpenSIPS in between). I also added the alias. But the > behaviour is as > before: > No notify message from freeswitch, neither after register nor > after a > voicemail is recorded. > > Best regards > Peter > Brian West schrieb: > > Yes an alias will be required for every domain you run on > the profile > > so it can find it. > > > > /b > > > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > > > > >> Try an alias on the sip profile. > >> > >> Mike > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Mailings at kh-dev.de Sat Dec 5 19:04:26 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 04:04:26 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Message-ID: Hi, currently I'm testing the newest FS trunk. Now I need a hint how to set up an "old" behavior of version 1.0.4. Here's the scenario: - Incoming call from caller_id_name: abc and caller_id_number: 123 - Now I set effective_caller_id_name: xyz and effective_caller_id_number: 456 - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) - After the pickup the Leg B is switching back to the "old" values and displays abc + 123 But I would rather see the new values during the call (as it was in version 1.0.4). What do I need to change? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/2ebc74d4/attachment-0002.html From yehavi.bourvine at gmail.com Sun Dec 6 00:12:35 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 10:12:35 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Message-ID: Hello Ognjen, From the tests I've done it is not so... When I set the profile to use INFO, and a phone calls and asks for RFC2833 (phone-events in the SDP) the FreeSwich ignores it (does not have phone-events field in the reply SDP) which causes the phone to not send RFC2833 events... Regards, __Yehavi: 2009/12/3 Ognjen Seslija > Bear in mind that FS will accept both 2833 and INFO in any profile on an > inbound call. Param "dtmf-type" is valid only for outbound calls from the > profile. > > Ognjen > > On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Hello, >> >> I have Polycom phones which send only RFC-2833 (or inband which I >> dislike) and they should go out to the PSTN via a Cisco gateway. The Cisco >> gateway has some bug and accepts only INFO. >> >> I did a few tests: >> >> - Some of the phones are on different profile than the Cisco. On their >> profile I set 'dtmf-type=rfc2833' and on the Cisco's profile I set >> 'dtmf-type=info' and Freeswitch did the translation. All works ok... >> - Some of the phones are on the same profile as the Cisco, so I must >> set dtmf-type to rfc2833; it works with internal applications (like >> voicemail) but does not work through the Cisco as it misinterprets the >> rfc2833 >> >> >> Is there a way to set some variable (or a parameter to the bridge >> application) to do the translation? >> >> Thanks! __Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/03f58458/attachment-0002.html From mailinglist at fribert.dk Sun Dec 6 01:24:49 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 10:24:49 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Message-ID: <4B1B8671020000E100000293@mail.fribert.dk> Hi Adam Excellent first steps! Thankyou for the hint. Now I hope somebody can tell me what I'm doing wrong next... I've gotten it to register to the testprovider here (musimi.dk), but I get an error when I create an account for testing with the X-Lite phone. It displays 403 forbidden in the display. I've created an account on FreeSwitch extension 1001 password 1001 mailbox 1001 voicemail password 1001 account code 1001 Effective Caller ID Name Fribert Effective Caller ID Number 4692xxxx (the Musimi number) Voicemail Mail To Voicemail Attach File true User Context default Call Group <> Enabled true Extension Description Test number In the X-Lite Display Name Fribert User name 1001 Password 1001 Autorization user name 1001 Domain LAN-IP-OF-pfSense Check in Register with domain and receive incoming calls Check in domain. That's about it. Looking on the status page, I can see these lines in the log: 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding 192.168.42.0/24 (deny) to list lan 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding 192.168.42.42/32 (allow) to list lan 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up listening on 0.0.0.0:8021 Which I'm kinda confused about, I don't have any 192.168 net here??? But as it also primarily forbids it, except .42 to allow, I'm wondering if it could be something internal? Best regards >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: I used the pfSense FreeSWITCH for awhile, as it is the only GUI FreeSWITCH I have found with a stable release. It was very easy to use, I would recommend it if you just want a quick base system with standard features. Though, I ended up switching to a compiled version of FreeSWITCH in order to make the customizations I needed for my office. http://doc.pfsense.org/index.php/FreeSWITCH -AF From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mailinglist Sent: Saturday, December 05, 2009 2:47 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? Has anybody done this? I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there's any help to be found on FreeSwitch? Anybody that can give pointers to a good step-by-step instruction? I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider. Of course transferring calls between the two, as well as group calls would be a nice benefit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/42fd5f2d/attachment-0002.html From freeswitch-users-list at metik.com Sun Dec 6 02:24:26 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 06 Dec 2009 05:24:26 -0500 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> Message-ID: <4B1B865A.2060901@metik.com> You previously stated that your Cisco gateway has some "bug" that prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on the voip dial-peer that the call is using? Unless you have configured the Cisco to support assymetric SDP or are using a non-default "rtp payload-type nte" setting that does not agree to well with FS's (default) "rfc2833-pt" setting, you should not have to use (SIP) INFO unless you want to. I would recommend doing the following to ensure you are hitting the correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) output: DTMF Relay = enabled, dtmf-relay = rtp-nte, example: show dialplan number 5551212 | i (dtmf-relay|DTMF Relay) DTMF Relay = enabled, dtmf-relay = rtp-nte, Also, you can sift through "show sip-ua calls" for the call and ensure that the value of "Negotiated Dtmf-relay" is "rtp-nte". -metik Yehavi Bourvine wrote: > Hello Ognjen, > > From the tests I've done it is not so... When I set the profile to > use INFO, and a phone calls and asks for RFC2833 (phone-events in the > SDP) the FreeSwich ignores it (does not have phone-events field in the > reply SDP) which causes the phone to not send RFC2833 events... > > Regards, __Yehavi: > > 2009/12/3 Ognjen Seslija > > > Bear in mind that FS will accept both 2833 and INFO in any profile > on an inbound call. Param "dtmf-type" is valid only for outbound > calls from the profile. > > Ognjen > > On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine > > wrote: > > Hello, > > I have Polycom phones which send only RFC-2833 (or inband > which I dislike) and they should go out to the PSTN via a > Cisco gateway. The Cisco gateway has some bug and accepts only > INFO. > > I did a few tests: > > * > Some of the phones are on different profile than the > Cisco. On their profile I set 'dtmf-type=rfc2833' and on > the Cisco's profile I set 'dtmf-type=info' and > Freeswitch did the translation. All works ok... > * > Some of the phones are on the same profile as the Cisco, > so I must set dtmf-type to rfc2833; it works with > internal applications (like voicemail) but does not work > through the Cisco as it misinterprets the rfc2833 > > > Is there a way to set some variable (or a parameter to the > bridge application) to do the translation? > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Sun Dec 6 03:59:01 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 13:59:01 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4B1B865A.2060901@metik.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> Message-ID: Hello Metik, 2009/12/6 Metik > You previously stated that your Cisco gateway has some "bug" that > prevents you from using RFC2833, did you enable "dtmf-relay rtp-nte" on > the voip dial-peer that the call is using? > > It is a PSTN dialpeer here, and it cannot be defined on it... > Unless you have configured the Cisco to support assymetric SDP or are > using a non-default "rtp payload-type nte" setting that does not agree > to well with FS's (default) "rfc2833-pt" setting, you should not have to > use (SIP) INFO unless you want to. > > I would recommend doing the following to ensure you are hitting the > correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): > > command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) > > Unfortunately this does not work on PSTN dial peers. > > Also, you can sift through "show sip-ua calls" for the call and ensure > that the value of "Negotiated Dtmf-relay" is "rtp-nte". > > This indeed shows that it has negotiated rtp-nte. Even when I do debug for CCAPI events (I think) I see it decodes the DTMFs; however, it ignores them while it accepts them via INFO. As I said: I guess this is a bug. Since the gateway is on a remote site I hesitate on upgrading it until I hae the chance to go there. Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/356baed7/attachment-0002.html From Prometheus001 at gmx.net Sun Dec 6 06:14:42 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 15:14:42 +0100 Subject: [Freeswitch-users] Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1B8671020000E100000293@mail.fribert.dk> References: <4B1B8671020000E100000293@mail.fribert.dk> Message-ID: <4B1BBC52.3050106@gmx.net> Concerning, > Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests. Can you set the console_log_level to "debug" in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb: > Hi Adam > > Excellent first steps! > Thankyou for the hint. > Now I hope somebody can tell me what I'm doing wrong next... > > I've gotten it to register to the testprovider here (musimi.dk), but I > get an error when I create an account for testing with the X-Lite phone. > > It displays 403 forbidden in the display. > > I've created an account on FreeSwitch > > extension 1001 > password 1001 > mailbox 1001 > voicemail password 1001 > account code 1001 > Effective Caller ID Name Fribert > Effective Caller ID Number 4692xxxx (the Musimi number) > Voicemail Mail To > Voicemail Attach File true > User Context default > Call Group <> > Enabled true > Extension Description Test number > > In the X-Lite > Display Name Fribert > User name 1001 > Password 1001 > Autorization user name 1001 > Domain LAN-IP-OF-pfSense > > Check in Register with domain and receive incoming calls > Check in domain. > > That's about it. > > Looking on the status page, I can see these lines in the log: > 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding > 192.168.42.0/24 (deny) to list lan > 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding > 192.168.42.42/32 (allow) to list lan > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > > Which I'm kinda confused about, I don't have any 192.168 net here??? > But as it also primarily forbids it, except .42 to allow, I'm > wondering if it could be something internal? > > Best regards > > > >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i > meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: > > I used the pfSense FreeSWITCH for awhile, as it is the only GUI > FreeSWITCH I have found with a stable release. It was very easy to > use, I would recommend it if you just want a quick base system with > standard features. Though, I ended up switching to a compiled version > of FreeSWITCH in order to make the customizations I needed for my office. > > http://doc.pfsense.org/index.php/FreeSWITCH > > -AF > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *mailinglist > *Sent:* Saturday, December 05, 2009 2:47 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a > pfSensefirewall??? > > > > Has anybody done this? > > > > I'm completely at a loss, having tinkered very little with Asterisk, > and giving up on that, I wonder if there's any help to be found on > FreeSwitch? > > Anybody that can give pointers to a good step-by-step instruction? > > > > I want to have it handle my two sip-phones (siemens dect ip and spa > 901), and handle a sip account at my provider. > > Of course transferring calls between the two, as well as group calls > would be a nice benefit. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jbr at consiglia.dk Sun Dec 6 06:22:41 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 6 Dec 2009 15:22:41 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: Dear all Some feedback regarding using Lua to access core database: First of all, I did not succeed to get SQLite drivers in Lua or ODBC-drivers in Lua to work. The SQLite driver did compile OK, but there was an error when loading into Lua. The ODBC driver did also compile OK, did load into Lua, but could not connect. Accessing the SQLite from the Linux console using "isql -v " worked OK. The problems may be related to the present Linux distribution, which is Ubuntu 9.1 server. Unfortunately the public searchable information about Lua ODBC driver problems is sparse. So I continued to try to get the FS to use MySQL as the core db. A number of problem occurred, which I did not find solution for in the FS documents. The problems and solutions are described below: 1) The core database is not automatically created by FS, therefore I created it manually. 2) During startup, the FS test for transaction support, and this test failed. To achieved transaction support with MySQL and MyODBC, three things had to be changed: a. A line was added in my.cnf to force innoDB as the default table: under the [mysqld] header, the following line was added: set-variable = default-table-type=InnoDB. b. The a line under the DNS was added to allow for multiple line statement support: option = 67108864. (ODBC version is 3.51). 3) After these changes the transaction worked, but all the tables in the core db were not created, therefore I copied the structure from the SQLite tables into tables with the same names in the MySQL database. This exercise also showed what the problem was: MySQL could not create tables with many VARCHAR type files with a size of 4096 (sound very big?). The size was reduced to 255, and most of the tables were created OK. One table still gave problems: the interface table. One of the fields is called key, which is a reserved word in MySQL, and by backticking the word key in the create statement, it worked. 4) Finally the FS started up using the MySQL, but errors splashed over the screen just after startup. There was a problem creating new records in the interface table, the problem was the key field. Changing the insert statement in switch.core.sqldb.c file by backticking the key field name and recompiling the FS solved that problem. I guess this will be fixed in later releases and I hope this will assist the brave programmers! I would like to argue for the development of SQLite connectivity in Lua. The ODBC core solution is not as clean as a direct database connection, and as long as this is limited to SQLite, a direct connection from "recommended script language" would be the cleanest solution. Further, it would be nice if everything works after having compiled the FS package. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/5c1f6c40/attachment-0002.html From Prometheus001 at gmx.net Sun Dec 6 08:22:30 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 17:22:30 +0100 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? Message-ID: <4B1BDA46.7010803@gmx.net> Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter From mailinglist at fribert.dk Sun Dec 6 10:13:02 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 19:13:02 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? Message-ID: <4B1C023E020000E1000002A2@mail.fribert.dk> Hi Peter Ok, I got the net changed in acl.conf.xml. I then tried setting console_loglevel, but I don't see any output on the console, it could very well be because it's a FreeBSD, and has very limited console. But after a restart it registers! So for some reason it needed a nudge there, very interesting. So now I have a local extension registered, and a provider registered, now I just need them to communicate. As I understand it, that's the dialplan I have to look at. I only have one provider hooked up, so a dial should be simple, right? It as Order 001 Condition ^0(.\d+)$ action bridge As I understand it, it should react on the 0 (for outside dialing) and then strip it And the action bridges the call to the outside. But I guess I'm missing something, because I just get a 'temporarily unavailable' shown in the xlite. >>> 06-12-2009 kl. 15:14 skrev Peter P GMX i meddelelsen <4B1BBC52.3050106 at gmx.net>: Concerning, > Which I'm kinda confused about, I don't have any 192.168 net here?? I think, this is a default entry in the acl.conf.xml. Please check the entries there. But normally this shouldn't stop freeswitch from working and handling requests. Can you set the console_log_level to "debug" in vars.xml and post you console output when the phone tries to register? You may also grep the network traffic on port 5060 (e.g. ngrep -d any port 5060 -W byline) on your machine, to see what's wrong. Best regards Peter mailinglist schrieb: > Hi Adam > > Excellent first steps! > Thankyou for the hint. > Now I hope somebody can tell me what I'm doing wrong next... > > I've gotten it to register to the testprovider here (musimi.dk), but I > get an error when I create an account for testing with the X-Lite phone. > > It displays 403 forbidden in the display. > > I've created an account on FreeSwitch > > extension 1001 > password 1001 > mailbox 1001 > voicemail password 1001 > account code 1001 > Effective Caller ID Name Fribert > Effective Caller ID Number 4692xxxx (the Musimi number) > Voicemail Mail To > Voicemail Attach File true > User Context default > Call Group <> > Enabled true > Extension Description Test number > > In the X-Lite > Display Name Fribert > User name 1001 > Password 1001 > Autorization user name 1001 > Domain LAN-IP-OF-pfSense > > Check in Register with domain and receive incoming calls > Check in domain. > > That's about it. > > Looking on the status page, I can see these lines in the log: > 2009-12-06 09:55:50.310829 [NOTICE] switch_core.c:1064 Adding > 192.168.42.0/24 (deny) to list lan > 2009-12-06 09:55:50.310842 [NOTICE] switch_core.c:1064 Adding > 192.168.42.42/32 (allow) to list lan > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > 2009-12-06 09:55:50.311256 [DEBUG] mod_event_socket.c:2291 Socket up > listening on 0.0.0.0:8021 > > Which I'm kinda confused about, I don't have any 192.168 net here??? > But as it also primarily forbids it, except .42 to allow, I'm > wondering if it could be something internal? > > Best regards > > > >>> 05-12-2009 kl. 18:52 skrev "Adam Ford" i > meddelelsen <0E0013F55E224674A1361329CF7A85F0 at redbonez>: > > I used the pfSense FreeSWITCH for awhile, as it is the only GUI > FreeSWITCH I have found with a stable release. It was very easy to > use, I would recommend it if you just want a quick base system with > standard features. Though, I ended up switching to a compiled version > of FreeSWITCH in order to make the customizations I needed for my office. > > http://doc.pfsense.org/index.php/FreeSWITCH > > -AF > > ------------------------------------------------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *mailinglist > *Sent:* Saturday, December 05, 2009 2:47 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Setting up a FreeSwitch system on a > pfSensefirewall??? > > > > Has anybody done this? > > > > I'm completely at a loss, having tinkered very little with Asterisk, > and giving up on that, I wonder if there's any help to be found on > FreeSwitch? > > Anybody that can give pointers to a good step-by-step instruction? > > > > I want to have it handle my two sip-phones (siemens dect ip and spa > 901), and handle a sip account at my provider. > > Of course transferring calls between the two, as well as group calls > would be a nice benefit. > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/f46a2acd/attachment-0002.html From lon at kickasspixels.com Sun Dec 6 10:13:01 2009 From: lon at kickasspixels.com (Lon Baker) Date: Sun, 6 Dec 2009 10:13:01 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> Jon, What version of MySQL are you using? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/d2f985d1/attachment-0002.html From Mailings at kh-dev.de Sun Dec 6 10:38:54 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 19:38:54 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: References: Message-ID: Hi, I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone with caller_id_name and caller_id_number instead of effective_caller_id_name and effective_caller_id_number values. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Klaus Hochlehnert Sent: Sunday, December 06, 2009 4:04 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Hi, currently I'm testing the newest FS trunk. Now I need a hint how to set up an "old" behavior of version 1.0.4. Here's the scenario: - Incoming call from caller_id_name: abc and caller_id_number: 123 - Now I set effective_caller_id_name: xyz and effective_caller_id_number: 456 - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) - After the pickup the Leg B is switching back to the "old" values and displays abc + 123 But I would rather see the new values during the call (as it was in version 1.0.4). What do I need to change? Thanks, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/a889bad5/attachment-0002.html From mrene_lists at avgs.ca Sun Dec 6 10:42:17 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 6 Dec 2009 13:42:17 -0500 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: References: Message-ID: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag > message to the phone with > caller_id_name and caller_id_number instead of > effective_caller_id_name and effective_caller_id_number values. > > Thanks, Klaus > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Klaus Hochlehnert > Sent: Sunday, December 06, 2009 4:04 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] caller_id_name + caller_id_number and > Snom 360... > > Hi, > > currently I?m testing the newest FS trunk. > Now I need a hint how to set up an ?old? behavior of version 1.0.4. > > Here?s the scenario: > - Incoming call from caller_id_name: abc and caller_id_number: 123 > - Now I set effective_caller_id_name: xyz and > effective_caller_id_number: 456 > - Leg B (Snom 360) is ringing and displays the new values (xyz + 456) > - After the pickup the Leg B is switching back to the ?old? values > and displays abc + 123 > > But I would rather see the new values during the call (as it was in > version 1.0.4). > What do I need to change? > > Thanks, Klaus > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/73d94204/attachment-0002.html From mailinglist at fribert.dk Sun Dec 6 11:06:45 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 20:06:45 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1C023E020000E1000002A2@mail.fribert.dk> References: <4B1C023E020000E1000002A2@mail.fribert.dk> Message-ID: <4B1C0ED5020000E1000002AC@mail.fribert.dk> Just got the freeswitch started from the command line, and got a bit of logging out I started with the 'multi homed' write up mentioned above (http://wiki.freeswitch.org/wiki/Multi_home_tutorial), as I of course have several nic's on it as it's a firewall. The reason I wanted to use the pfSense firewall, is because I'll get rid of the NAT'in imposed by having it on a local machine. My main reason for setting something up at the first place was to get some of my very limited external IP's back. I've changed the conf/sip_profiles/internal.xml to reflect my LAN addresses My lan is 10.11.12.x My wan is 87.61.18.196 If I start it, it shows a lot on the screen, a line in red is scrolled right out of there, and the buffer isn't large enough to go back 2009-12-06 19:35:42.928556 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 3 0ms 2009-12-06 19:35:42.928564 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 4 0ms 2009-12-06 19:35:42.928573 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 5 0ms 2009-12-06 19:35:42.928583 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 6 0ms 2009-12-06 19:35:42.928592 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 7 0ms 2009-12-06 19:35:42.928600 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 8 0ms 2009-12-06 19:35:42.928609 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 9 0ms 2009-12-06 19:35:42.928619 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 00ms 2009-12-06 19:35:42.928628 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 10ms 2009-12-06 19:35:42.928636 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G729' (G.729) 8000hz 1 20ms 2009-12-06 19:35:42.928922 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_amr] 2009-12-06 19:35:42.928937 [NOTICE] switch_loadable_module.c:182 Adding Codec 'AMR' (AMR) 8000hz 20ms 2009-12-06 19:35:42.930956 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_ilbc] 2009-12-06 19:35:42.930999 [NOTICE] switch_loadable_module.c:182 Adding Codec 'iLBC' (iLBC) 8000hz 30 ms 2009-12-06 19:35:42.931014 [NOTICE] switch_loadable_module.c:182 Adding Codec 'iLBC' (iLBC) 8000hz 20 ms 2009-12-06 19:35:42.938433 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_speex] 2009-12-06 19:35:42.938466 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2009-12-06 19:35:42.938908 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2009-12-06 19:35:42.938931 [NOTICE] switch_loadable_module.c:182 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2009-12-06 19:35:42.939735 [INFO] mod_siren.c:141 Audio coding: ITU-T Rec. G.722.1, licensed from Pol ycom(R) 2009-12-06 19:35:42.939766 [INFO] mod_siren.c:142 Audio coding: ITU-T Rec. G.722.1 Annex C, licensed from Polycom(R) 2009-12-06 19:35:42.939794 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_siren] 2009-12-06 19:35:42.939810 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 20ms 2009-12-06 19:35:42.939824 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 40ms 2009-12-06 19:35:42.939834 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 32000hz 60ms 2009-12-06 19:35:42.939843 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 20ms 2009-12-06 19:35:42.939853 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 40ms 2009-12-06 19:35:42.939862 [NOTICE] switch_loadable_module.c:182 Adding Codec 'G7221' (Polycom(R) G72 2.1/G722.1C) 16000hz 60ms 2009-12-06 19:35:42.950262 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_celt] 2009-12-06 19:35:42.950297 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 10ms 2009-12-06 19:35:42.950311 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 8ms 2009-12-06 19:35:42.950322 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 6ms 2009-12-06 19:35:42.950333 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 4ms 2009-12-06 19:35:42.950342 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 48000hz 2ms 2009-12-06 19:35:42.950351 [NOTICE] switch_loadable_module.c:182 Adding Codec 'CELT' (CELT ultra-low delay) 32000hz 10ms 2009-12-06 19:35:42.950804 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_h26x] 2009-12-06 19:35:42.950822 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H264' (H.264 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.950912 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263' (H.263 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.950991 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263-1998' (H.263+ Vid eo (passthru)) 90000hz 0ms 2009-12-06 19:35:42.951072 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H263-2000' (H.263++ Vi deo (passthru)) 90000hz 0ms 2009-12-06 19:35:42.951189 [NOTICE] switch_loadable_module.c:182 Adding Codec 'H261' (H.261 Video (pa ssthru)) 90000hz 0ms 2009-12-06 19:35:42.966090 [INFO] mod_sndfile.c:330 LibSndFile Version : libsndfile-1.0.19 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") WVE (Psion Series 3) (extension "wve") XI (FastTracker 2) (extension "xi") ================================================================================ 2009-12-06 19:35:42.966344 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_sndfile] 2009-12-06 19:35:42.966358 [NOTICE] switch_loadable_module.c:294 Adding File Format 'aiff' 2009-12-06 19:35:42.966500 [NOTICE] switch_loadable_module.c:294 Adding File Format 'au' 2009-12-06 19:35:42.966594 [NOTICE] switch_loadable_module.c:294 Adding File Format 'avr' 2009-12-06 19:35:42.966672 [NOTICE] switch_loadable_module.c:294 Adding File Format 'caf' 2009-12-06 19:35:42.966747 [NOTICE] switch_loadable_module.c:294 Adding File Format 'htk' 2009-12-06 19:35:42.966820 [NOTICE] switch_loadable_module.c:294 Adding File Format 'iff' 2009-12-06 19:35:42.966894 [NOTICE] switch_loadable_module.c:294 Adding File Format 'mat' 2009-12-06 19:35:42.966998 [NOTICE] switch_loadable_module.c:294 Adding File Format 'paf' 2009-12-06 19:35:42.967074 [NOTICE] switch_loadable_module.c:294 Adding File Format 'pvf' 2009-12-06 19:35:42.967148 [NOTICE] switch_loadable_module.c:294 Adding File Format 'raw' 2009-12-06 19:35:42.968115 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sd2' 2009-12-06 19:35:42.968226 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sds' 2009-12-06 19:35:42.968304 [NOTICE] switch_loadable_module.c:294 Adding File Format 'sf' 2009-12-06 19:35:42.968379 [NOTICE] switch_loadable_module.c:294 Adding File Format 'voc' 2009-12-06 19:35:42.968453 [NOTICE] switch_loadable_module.c:294 Adding File Format 'w64' 2009-12-06 19:35:42.968527 [NOTICE] switch_loadable_module.c:294 Adding File Format 'wav' 2009-12-06 19:35:42.968601 [NOTICE] switch_loadable_module.c:294 Adding File Format 'wve' 2009-12-06 19:35:42.968677 [NOTICE] switch_loadable_module.c:294 Adding File Format 'xi' 2009-12-06 19:35:42.968751 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r8' 2009-12-06 19:35:42.968825 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r16' 2009-12-06 19:35:42.968899 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r24' 2009-12-06 19:35:42.968973 [NOTICE] switch_loadable_module.c:294 Adding File Format 'r32' 2009-12-06 19:35:42.969047 [NOTICE] switch_loadable_module.c:294 Adding File Format 'gsm' 2009-12-06 19:35:42.969121 [NOTICE] switch_loadable_module.c:294 Adding File Format 'ul' 2009-12-06 19:35:42.969196 [NOTICE] switch_loadable_module.c:294 Adding File Format 'al' 2009-12-06 19:35:42.969271 [NOTICE] switch_loadable_module.c:294 Adding File Format 'adpcm' 2009-12-06 19:35:42.969751 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_native_fil e] 2009-12-06 19:35:42.969772 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263' 2009-12-06 19:35:42.969886 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G7221' 2009-12-06 19:35:42.969969 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AMR' 2009-12-06 19:35:42.970045 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263-1998' 2009-12-06 19:35:42.970122 [NOTICE] switch_loadable_module.c:294 Adding File Format 'SPEEX' 2009-12-06 19:35:42.970197 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G729' 2009-12-06 19:35:42.970274 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G723' 2009-12-06 19:35:42.970352 [NOTICE] switch_loadable_module.c:294 Adding File Format 'LPC' 2009-12-06 19:35:42.970427 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-16' 2009-12-06 19:35:42.970502 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H261' 2009-12-06 19:35:42.970577 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-16' 2009-12-06 19:35:42.970653 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PCMA' 2009-12-06 19:35:42.970729 [NOTICE] switch_loadable_module.c:294 Adding File Format 'DVI4' 2009-12-06 19:35:42.970804 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-24' 2009-12-06 19:35:42.970880 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PCMU' 2009-12-06 19:35:42.970956 [NOTICE] switch_loadable_module.c:294 Adding File Format 'L16' 2009-12-06 19:35:42.971031 [NOTICE] switch_loadable_module.c:294 Adding File Format 'iLBC' 2009-12-06 19:35:42.971106 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PROXY' 2009-12-06 19:35:42.971217 [NOTICE] switch_loadable_module.c:294 Adding File Format 'PROXY-VID' 2009-12-06 19:35:42.971294 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-24' 2009-12-06 19:35:42.971369 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-32' 2009-12-06 19:35:42.971463 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H263-2000' 2009-12-06 19:35:42.971540 [NOTICE] switch_loadable_module.c:294 Adding File Format 'H264' 2009-12-06 19:35:42.971616 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-32' 2009-12-06 19:35:42.971691 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G722' 2009-12-06 19:35:42.971767 [NOTICE] switch_loadable_module.c:294 Adding File Format 'CELT' 2009-12-06 19:35:42.971843 [NOTICE] switch_loadable_module.c:294 Adding File Format 'AAL2-G726-40' 2009-12-06 19:35:42.971918 [NOTICE] switch_loadable_module.c:294 Adding File Format 'G726-40' 2009-12-06 19:35:42.971993 [NOTICE] switch_loadable_module.c:294 Adding File Format 'GSM' 2009-12-06 19:35:42.973983 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_local_stre am] 2009-12-06 19:35:42.974028 [NOTICE] switch_loadable_module.c:270 Adding API Function 'stop_local_stre am' 2009-12-06 19:35:42.975627 [CONSOLE] mod_local_stream.c:142 Can't open directory: /usr/local/freeswit ch/sounds/music/16000 2009-12-06 19:35:42.975972 [CONSOLE] mod_local_stream.c:142 Can't open directory: /usr/local/freeswit ch/sounds/music/32000 2009-12-06 19:35:42.976108 [NOTICE] switch_loadable_module.c:270 Adding API Function 'start_local_str eam' 2009-12-06 19:35:42.976214 [NOTICE] switch_loadable_module.c:270 Adding API Function 'show_local_stre am' 2009-12-06 19:35:42.976297 [NOTICE] switch_loadable_module.c:294 Adding File Format 'local_stream' 2009-12-06 19:35:42.976779 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_tone_strea m] 2009-12-06 19:35:42.976799 [NOTICE] switch_loadable_module.c:294 Adding File Format 'tone_stream' 2009-12-06 19:35:42.976892 [NOTICE] switch_loadable_module.c:294 Adding File Format 'silence_stream' 2009-12-06 19:35:43.96229 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_teletone.so] 2009-12-06 19:35:43.96554 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_core_db.so] 2009-12-06 19:35:43.96828 [CONSOLE] mod_spidermonkey.c:947 Successfully Loaded [/usr/local/freeswitch /mod/mod_spidermonkey_socket.so] 2009-12-06 19:35:43.102288 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_spidermonk ey] 2009-12-06 19:35:43.102327 [NOTICE] switch_loadable_module.c:248 Adding Application 'javascript' 2009-12-06 19:35:43.102515 [NOTICE] switch_loadable_module.c:270 Adding API Function 'jsrun' 2009-12-06 19:35:43.102601 [NOTICE] switch_loadable_module.c:270 Adding API Function 'jsapi' 2009-12-06 19:35:43.110060 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_lua] 2009-12-06 19:35:43.110079 [NOTICE] switch_loadable_module.c:248 Adding Application 'lua' 2009-12-06 19:35:43.110252 [NOTICE] switch_loadable_module.c:270 Adding API Function 'luarun' 2009-12-06 19:35:43.110338 [NOTICE] switch_loadable_module.c:270 Adding API Function 'lua' 2009-12-06 19:35:43.110761 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_say_en] 2009-12-06 19:35:43.110780 [NOTICE] switch_loadable_module.c:395 Adding Say interface 'en' 2009-12-06 19:35:43.110881 [CONSOLE] switch_loadable_module.c:120 Starting runtime thread for CORE_SO FTTIMER_MODULE 2009-12-06 19:35:43.111049 [CONSOLE] switch_loadable_module.c:120 Starting runtime thread for mod_eve nt_socket 2009-12-06 19:35:43.111255 [NOTICE] switch_core.c:898 Created ip list rfc1918.auto default (deny) 2009-12-06 19:35:43.111279 [NOTICE] switch_core.c:906 Created ip list nat.auto default (deny) 2009-12-06 19:35:43.111293 [NOTICE] switch_core.c:914 Created ip list loopback.auto default (deny) 2009-12-06 19:35:43.111348 [NOTICE] switch_core.c:920 Created ip list localnet.auto default (deny) 2009-12-06 19:35:43.111361 [NOTICE] switch_core.c:923 Adding 87.61.18.196/255.255.255.248 (allow) to list localnet.auto 2009-12-06 19:35:43.111380 [CONSOLE] switch_core.c:961 Created ip list lan default (allow) 2009-12-06 19:35:43.111395 [NOTICE] switch_core.c:1064 Adding 10.11.12.0/24 (deny) to list lan 2009-12-06 19:35:43.111422 [NOTICE] switch_core.c:1064 Adding 10.11.12.25/32 (allow) to list lan 2009-12-06 19:35:43.111432 [CONSOLE] switch_core.c:961 Created ip list domains default (deny) 2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 2009-12-06 19:35:43.111986 [CONSOLE] switch_core.c:1465 FreeSWITCH Version 1.0.trunk (13784) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at firewall.fribert.dk> If I then inquire about the internal status I get: freeswitch at firewall.fribert.dk> sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia at 10.11.12.25:5060 BIND-URL sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= If I do a external status it shows: freeswitch at firewall.fribert.dk> sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia at 87.61.18.196:5080 BIND-URL sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= ================================================================================================= As far as I can see, everything looks ok, except for the 2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 I'm wondering WHY it wants a domain on the external IP??? I then started the SIP softphone, and got: 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute and you must configure your device to use the proper domain in it's authentication credentials. 2009-12-06 19:36:27.988290 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute and you must configure your device to use the proper domain in it's authentication credentials. I've set up the phone to use a domain 10.11.12.25 But I guess, something is screwy with the 'domain' definition as it shows above. So now I'm halted. P.S. I'm full of aw, of the quick and qualified help I've gotten so far, with good pointers to where to start. Thankyou all for your input! This is a brand new line of work for me, so I'm completely at a loss, which is quite fun when I'm used to be the server expert :-D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/d83c0702/attachment-0002.html From freeswitch-users-list at metik.com Sun Dec 6 11:16:43 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 06 Dec 2009 14:16:43 -0500 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> Message-ID: <4B1C031B.8060906@metik.com> Unless the IOS you are running is extremely buggy, "debug voip ccapi" commands should not provide you with that detail, what you really want to use is "debug voip rtp session named-event". Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik > > > You previously stated that your Cisco gateway has some "bug" that > prevents you from using RFC2833, did you enable "dtmf-relay > rtp-nte" on > the voip dial-peer that the call is using? > > > It is a PSTN dialpeer here, and it cannot be defined on it... > > > Unless you have configured the Cisco to support assymetric SDP or are > using a non-default "rtp payload-type nte" setting that does not agree > to well with FS's (default) "rfc2833-pt" setting, you should not > have to > use (SIP) INFO unless you want to. > > I would recommend doing the following to ensure you are hitting the > correct dial-peer and it is configured for RFC 2833 ("rtp-nte"): > > command: show dialplan number [number] | i (dtmf-relay|DTMF Relay) > > > Unfortunately this does not work on PSTN dial peers. > > > > Also, you can sift through "show sip-ua calls" for the call and ensure > that the value of "Negotiated Dtmf-relay" is "rtp-nte". > > > This indeed shows that it has negotiated rtp-nte. Even when I do debug > for CCAPI events (I think) I see it decodes the DTMFs; however, it > ignores them while it accepts them via INFO. As I said: I guess this > is a bug. > > Since the gateway is on a remote site I hesitate on upgrading it until > I hae the chance to go there. > > Thanks, __Yehavi: > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Sun Dec 6 11:29:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:29:14 -0600 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? In-Reply-To: <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> References: <4B1BDA46.7010803@gmx.net> <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> Message-ID: <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> Someone else was asking about this too. I could probably write a dictaction mod in c like the one I made for asterisk starting at about $3k depending on the featureset required. On Dec 6, 2009 10:30 AM, "Peter P GMX" wrote: Hello, I would like to offer a dictation service to a secretary. Means: * the boss is dictating some text on a certain phone number * the secretary picks up the recording on the phone and types the text into the computer As the secretary is not able to type in as fastly as heir boss is able to speak, she needs some kind of pause and rewind button. 1st question: Is there any functionality available for example in uuid_broadcast? 2nd question: How much would be the effort to implement this (uuid_broadcast_pause, uuid_broadcast_UNpause, uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. Best regards Peter _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/4e85c5cb/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 6 11:32:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:32:17 -0600 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> References: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> Message-ID: <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> Or set it to true depending on the case Also consider using set_profile_var to set the caller id explicitly instead of using effective. There is also effective_callee_id name and number you could set on the a leg. You'll have to expirement but the one mathieu said is your best bet. On Dec 6, 2009 12:47 PM, "Mathieu Rene" wrote: Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/59976acd/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 6 11:35:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:35:30 -0600 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <4B1C031B.8060906@metik.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> <4B1C031B.8060906@metik.com> Message-ID: <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> Some more bad news for you, info dtmf spec has expired and has been abandoned. Wait till you see what they did accept instead...... On Dec 6, 2009 1:22 PM, "Metik" wrote: Unless the IOS you are running is extremely buggy, "debug voip ccapi" commands should not provide you with that detail, what you really want to use is "debug voip rtp session named-event". Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik < freeswitch-users-list at metik.com > > > > You previously stated that your Cisco gateway has some "bug" that > prevents you from us... > ------------------------------------------------------------------------ > > _____________________... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/6965b7f2/attachment-0002.html From yehavi.bourvine at gmail.com Sun Dec 6 11:36:02 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 6 Dec 2009 21:36:02 +0200 Subject: [Freeswitch-users] A few questions about Polycom setup Message-ID: Hello, I have a few questions about Ploycom's usage and provisioning for which I found no answers neither at the docs nor on the WEB: - I would like to enable SIP/TLS. for this I have to import the root certificate. How can I do it via the XML config files? the only method I found is via the phone's interface, but what do you do when you have tens and more of them? - Since the phone is limited to 3way conference I would like it to use a conference room on the server. I've defined: - The result is that when A calls B (the polycom phone) which tries to conference with C is that B does a conference with C and the conference room and A is left on hold... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/c1359a25/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 6 11:41:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 13:41:38 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> Message-ID: <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> Most of this is unfortunatly because you do not have the proper skill to set it up because, with the proper skills, all of the ways you tried would have ended sucessfully. I say that beacause I have had many users use each of the different methods in your list of failures only they were sucessful. What you are asking for is possible but would require many hours of coding just to help solve your problem. You would have to wait a really long time until someone had the time to do it for free or post a bounty for it. Probably about 1k in consulting time. It may be cheaper for you to pay a consultant to set up one of the ways known to work. These are your options as I see it. On Dec 6, 2009 12:20 PM, "Lon Baker" wrote: Jon, What version of MySQL are you using? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/ddd32fbb/attachment-0002.html From Mailings at kh-dev.de Sun Dec 6 12:37:30 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 6 Dec 2009 21:37:30 +0100 Subject: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... In-Reply-To: <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> References: <4056E855-EE90-465F-8CB2-564EF48D53D6@avgs.ca> <191c3a030912061132p5dfd6458n77d82ec4e1e0d121@mail.gmail.com> Message-ID: Ok, set_profile_var did the trick and also works with intercepted calls. Thanks, Klaus From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, December 06, 2009 8:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] caller_id_name + caller_id_number and Snom 360... Or set it to true depending on the case Also consider using set_profile_var to set the caller id explicitly instead of using effective. There is also effective_callee_id name and number you could set on the a leg. You'll have to expirement but the one mathieu said is your best bet. On Dec 6, 2009 12:47 PM, "Mathieu Rene" > wrote: Hi Klaus, Try setting ignore_display_updates=false Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Dec-09, at 1:38 PM, Klaus Hochlehnert wrote: > Hi, > > I just checked the SIP traces and it looks like FS sends a sipfrag message to the phone ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/458f5e89/attachment-0002.html From JCasale at activenetwerx.com Sun Dec 6 13:01:23 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 6 Dec 2009 21:01:23 +0000 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: <4B1C0ED5020000E1000002AC@mail.fribert.dk> References: <4B1C023E020000E1000002A2@mail.fribert.dk> <4B1C0ED5020000E1000002AC@mail.fribert.dk> Message-ID: >Registrations: >================================================================================================= >================================================================================================= >As far as I can see, everything looks ok, except for the >2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 >I'm wondering WHY it wants a domain on the external IP??? >? >? >I then started the SIP softphone, and got: >? >?2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] >You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute >and you must configure your device to use the proper domain in it's authentication credentials. Yea, it looks like your server is taking the domain of the wan nic. I don't begin to claim I know all there is to know about this (still lurking while I learn as well...) but I got a lab'ed up pfSense box to work only after I edited vars.xml and set: Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something to do with nat and clients in the lan accessing the wan ip. From jbr at consiglia.dk Sun Dec 6 13:33:33 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 6 Dec 2009 22:33:33 +0100 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> Message-ID: The MySQL version is 5.1.37. Well I'm not an expert on every field, and I have no skills in the C, include libraries, and the art of compiling. For this I have to follow the guidelines. But it wouldn't harm the FS project if it generally became more accessible to the race of non-specialists, which I hereby represent. Jon Br?el ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 6. december 2009 20:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db Most of this is unfortunatly because you do not have the proper skill to set it up because, with the proper skills, all of the ways you tried would have ended sucessfully. I say that beacause I have had many users use each of the different methods in your list of failures only they were sucessful. What you are asking for is possible but would require many hours of coding just to help solve your problem. You would have to wait a really long time until someone had the time to do it for free or post a bounty for it. Probably about 1k in consulting time. It may be cheaper for you to pay a consultant to set up one of the ways known to work. These are your options as I see it. On Dec 6, 2009 12:20 PM, "Lon Baker" > wrote: Jon, What version of MySQL are you using? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/bf4d00a1/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 6 14:12:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 6 Dec 2009 16:12:42 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912061407k3b0e6d68p244f25caf1add4bb@mail.gmail.com> References: <5d3e0dc60912061013j53482717m8acbac1911629550@mail.gmail.com> <191c3a030912061141m350bc172vc90a382471c65132@mail.gmail.com> <191c3a030912061358r79013c9ay445fca25dc24e054@mail.gmail.com> <191c3a030912061400s5994537et178bf2002de06b58@mail.gmail.com> <191c3a030912061405n689a80a5ube645b738db0b531@mail.gmail.com> <191c3a030912061407k3b0e6d68p244f25caf1add4bb@mail.gmail.com> Message-ID: <191c3a030912061412i13349f14n167ffa3dcd873021@mail.gmail.com> Yes, exactly my point. Like I said you have several choices.... be paitent till we have time to code it for free, post a bounty to increase the chance somone will do it from the community, hire someone to set it up for you or keep trying yourself. Did I miss something? On Dec 6, 2009 3:38 PM, "Jon Bruel" wrote: The MySQL version is 5.1.37. Well I?m not an expert on every field, and I have no skills in the C, include libraries, and the art of compiling. For this I have to follow the guidelines. But it wouldn?t harm the FS project if it generally became more accessible to the race of non-specialists, which I hereby represent. *Jon Br?el* ------------------------------ *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* 6. december 2009 20:42 *To:* freeswitch-users at lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Lua and database access to core_db Most of this is unfortunatly because you do not have the proper skill to set it up because, wit... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/9be08f45/attachment-0002.html From Prometheus001 at gmx.net Sun Dec 6 14:12:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sun, 06 Dec 2009 23:12:51 +0100 Subject: [Freeswitch-users] uuid_broadcast with pause and rewind for dictation service? In-Reply-To: <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> References: <4B1BDA46.7010803@gmx.net> <191c3a030912061128h5442ff21y63d3f29ad8595ba@mail.gmail.com> <191c3a030912061129q3b9b3b9cibfc080c71424060c@mail.gmail.com> Message-ID: <4B1C2C63.9080401@gmx.net> Oh that's a lot of money, anybody else needs this feature, so we may share a bounty? Best regards Peter Anthony Minessale schrieb: > > Someone else was asking about this too. > I could probably write a dictaction mod in c like the one I made for > asterisk starting at about $3k depending on the featureset required. > >> On Dec 6, 2009 10:30 AM, "Peter P GMX" > > wrote: >> >> Hello, >> >> I would like to offer a dictation service to a secretary. >> Means: >> >> * the boss is dictating some text on a certain phone number >> * the secretary picks up the recording on the phone and types the >> text into the computer >> >> As the secretary is not able to type in as fastly as heir boss is able >> to speak, she needs some kind of pause and rewind button. >> 1st question: Is there any functionality available for example in >> uuid_broadcast? >> 2nd question: How much would be the effort to implement this >> (uuid_broadcast_pause, uuid_broadcast_UNpause, >> uuid_broadcast_rewind(sec)) ? I may offer then a bounty for this. >> >> Best regards >> Peter >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mailinglist at fribert.dk Sun Dec 6 14:53:25 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sun, 06 Dec 2009 23:53:25 +0100 Subject: [Freeswitch-users] Svar: Re: Setting up a FreeSwitch system on a pfSensefirewall??? In-Reply-To: References: <4B1C023E020000E1000002A2@mail.fribert.dk> <4B1C0ED5020000E1000002AC@mail.fribert.dk> Message-ID: <4B1C43F5020000E1000002B1@mail.fribert.dk> Hi Joseph Ahh, yes, that got rid of that error :-) Now on to the next one. So now it's connecting, both at my provider, and my softphone. Now I have to figure out why it tells me 'Call failed: not found' when I try to call out of the system... But I think that's a task for tomorrow when I'm more awake :-D Thanks! Fribert >>> 06-12-2009 kl. 22:01 skrev "Joseph L. Casale" i meddelelsen : >Registrations: >================================================================================================= >================================================================================================= >As far as I can see, everything looks ok, except for the >2009-12-06 19:35:43.111477 [WARNING] switch_core.c:990 Cannot locate domain 87.61.18.196 >I'm wondering WHY it wants a domain on the external IP??? > > >I then started the SIP softphone, and got: > > 2009-12-06 19:36:23.588241 [WARNING] sofia_reg.c:1755 Can't find user [1001 at 87.61.18.196] >You must define a domain called '87.61.18.196' in your directory and add a user with the id="1001" at tribute >and you must configure your device to use the proper domain in it's authentication credentials. Yea, it looks like your server is taking the domain of the wan nic. I don't begin to claim I know all there is to know about this (still lurking while I learn as well...) but I got a lab'ed up pfSense box to work only after I edited vars.xml and set: Where 10.0.0.1 was the ip my internal.xml bound to. I assumed it had something to do with nat and clients in the lan accessing the wan ip. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/b9ad9246/attachment-0002.html From sklein at singular.com Sat Dec 5 19:21:16 2009 From: sklein at singular.com (Steve Klein) Date: Sat, 5 Dec 2009 19:21:16 -0800 Subject: [Freeswitch-users] lua+sqlite example? Message-ID: <04a201ca7623$2c0b2020$84216060$@com> Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091205/51e80e09/attachment-0002.html From sklein at singular.com Sun Dec 6 12:07:39 2009 From: sklein at singular.com (Steve Klein) Date: Sun, 6 Dec 2009 12:07:39 -0800 Subject: [Freeswitch-users] Database suggestions/pointers/? Message-ID: <053d01ca76af$c311d2c0$49357840$@com> Greetings. We need to add database access to an IVR application we are prototyping. Based on FS "best practice" suggestions, we are using Lua for the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite would be a recommended approach. However, we can't find any examples of this combo anywhere. So, what is the "best practice" scripting + database recommendation for a high-volume database-driven FS app? Any help/pointers greatly appreciated! --Steve Klein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091206/3261165c/attachment-0002.html From timuckun at gmail.com Sun Dec 6 16:59:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 7 Dec 2009 13:59:31 +1300 Subject: [Freeswitch-users] Database suggestions/pointers/? In-Reply-To: <053d01ca76af$c311d2c0$49357840$@com> References: <053d01ca76af$c311d2c0$49357840$@com> Message-ID: <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein wrote: > Greetings. We need to add database access to an IVR application we are > prototyping. Based on FS ?best practice? suggestions, we are using Lua for > the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite > would be a recommended approach. However, we can?t find any examples of this > combo anywhere. So, what is the ?best practice? scripting + database > recommendation for a high-volume database-driven FS app? > I would suggest you take a look at freeswitcher (http://github.com/bougyman/freeswitcher). The good thing is that it's ruby and therefore you can use any database compatible with ruby (that's all of them pretty much). You can also use an ORM of your choice or if you don't want to use an ORM you can use the amazingly fantastic sequel library. Being ruby it will run outside of the freeswitch memory space and you will have to use the inbound/outbound socket API. That may be a good thing if you want to separate your database and IVR logic from the machine running your freeswitch. Ruby is pretty easy to pick up if you don't know it and there are a wealth of libraries if you want to do other things like connect to web sites, manipulate XML, etc. There is also a liverpie http://github.com/jsgoecke/liverpie which is more of a proxy thing you can interface with any language. I am sure lua is nice but it seems like people are having some problems with ODBC, memory leaks etc when it comes to databases. If you go a ruby library that all goes away. From djbinter at yahoo.com Sun Dec 6 17:17:14 2009 From: djbinter at yahoo.com (DJB) Date: Sun, 6 Dec 2009 17:17:14 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. Message-ID: <168319.49226.qm@web37502.mail.mud.yahoo.com> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -------------------------------------------------------------------------------------------------------------- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? S References: <41A44DD027064988A914974405788C2E@procent> <191c3a030910050731m2d74979ep4598e5a1945d58ae@mail.gmail.com> <1254901192035-3780245.post@n2.nabble.com> <8437F5BC-7AFF-4A74-B8CD-C5B8219021F6@jerris.com> <1255008427639-3788019.post@n2.nabble.com> <191c3a030910080823g79c7c596x1cd887e1538ce2e1@mail.gmail.com> <1255169044209-3799274.post@n2.nabble.com> <1255337256919-3806786.post@n2.nabble.com> <59F3CD44-5FEA-403C-98BE-EEE49EC3815B@freeswitch.org> <1255363492193-3808860.post@n2.nabble.com> Message-ID: This bug has been now closed out in jira due to no response for requested information. If you wish to resolve this issue please follow up on your bugs when information is requested. Mike On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote: > > Nope, I wanted to make sure that this is indeed a bug. I opened an issue in > JIRA before regarding some other matter and it turned out to be my mistake, > so I decided to try mailing list first this time. > MA > > > > Brian West wrote: >> >> Did you open a jira and attach all the info? >> >> /b >> >> On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote: >> >>> Yes, I confirmed that with Wireshark (filter "rtp and ip.src == >>> ). RTP packets are sent every 20ms. >>> >>> MAniserowicz From jaybinks at gmail.com Sun Dec 6 20:04:03 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 7 Dec 2009 14:04:03 +1000 Subject: [Freeswitch-users] Audiocodes PRI Gateway Message-ID: Guys, im after info from people with experience with AudioCodes Mediant 2k PRI Gateways. specifically how well they inter-op with Freeswitch, and how compliant their SIP stack is. I guess the bottom line is, would you recommend these gateways or would you suggest something else ? -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/92522e65/attachment-0002.html From dujinfang at gmail.com Sun Dec 6 20:45:54 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Dec 2009 12:45:54 +0800 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario Message-ID: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 Thanks. From imthiyazg at gmail.com Sun Dec 6 21:52:46 2009 From: imthiyazg at gmail.com (Imthiyaz Ahmed) Date: Mon, 7 Dec 2009 11:22:46 +0530 Subject: [Freeswitch-users] Audiocodes PRI Gateway In-Reply-To: References: Message-ID: <8595daf70912062152v122d3acdvcd84db8162384ee4@mail.gmail.com> We are using Audiocodes and Sangoma netborder express GW with Freeswitch . it works well. Thanks Imthiyaz On Mon, Dec 7, 2009 at 9:34 AM, jay binks wrote: > Guys, > ??im after info from people with experience with AudioCodes Mediant 2k PRI > Gateways. > specifically?how well they?inter-op?with Freeswitch, and how compliant their > SIP stack is. > I guess the bottom line is, would you?recommend?these gateways or would you > suggest something else ? > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best Regards G.Imthiyaz Ahmed PeopleTech systems (P) ltd http://peopletech.co.in From mike at jerris.com Sun Dec 6 21:53:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 7 Dec 2009 00:53:33 -0500 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario In-Reply-To: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> References: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> Message-ID: <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> Please report bugs to jira.freeswitch.org. Mike On Dec 6, 2009, at 11:45 PM, Seven Du wrote: > Hi, > > I know there's some chang on att_xfer, and after upgrade(re-bootstrap) > to trunk code, no sound after att_xfer. > > Then I rebuild FS 15807 with a fresh checkout, but still using the old > conf/ settings, sound is ok, but there are other problems: > > A call B, and B att_xfer C > > 1) origination_cancel_key not working. no even no DTMF log in FS when > I press # or any other key, I tried with Zoiper and Snom(on the B leg) > 2) when C answers, B immediately hangup, so B has no chance talk to C > > Could this be a problem? I pasted logs: > > http://pastebin.freeswitch.org/11417 From abeka at greatiam.com Sun Dec 6 23:30:45 2009 From: abeka at greatiam.com (Samuel Abekah-Mensah) Date: Mon, 07 Dec 2009 07:30:45 +0000 Subject: [Freeswitch-users] Mutual Registration of servers Message-ID: <4B1CAF25.6010706@greatiam.com> Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. From dujinfang at gmail.com Mon Dec 7 02:21:29 2009 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Dec 2009 18:21:29 +0800 Subject: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario In-Reply-To: <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> References: <23f91030912062045k70d9b5b9h4d9aea9abf556bed@mail.gmail.com> <11E0EFD2-7924-4E86-8486-3951BDA0DBF2@jerris.com> Message-ID: <23f91030912070221v3be54ed5x2a1ac1e6c9bf0e7d@mail.gmail.com> Thanks, done. 2009/12/7 Michael Jerris : > Please report bugs to jira.freeswitch.org. > > Mike > > On Dec 6, 2009, at 11:45 PM, Seven Du wrote: > >> Hi, >> >> I know there's some chang on att_xfer, and after upgrade(re-bootstrap) >> to trunk code, no sound after att_xfer. >> >> Then I rebuild FS 15807 with a fresh checkout, but still using the old >> conf/ settings, sound is ok, but there are other problems: >> >> A call B, and B att_xfer C >> >> 1) origination_cancel_key not working. no even no DTMF log in FS when >> I press # or any other key, I tried with Zoiper and Snom(on the B leg) >> 2) when C answers, B immediately hangup, so B has no chance talk to C >> >> Could this be a problem? I pasted logs: >> >> http://pastebin.freeswitch.org/11417 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lakindia89 at gmail.com Mon Dec 7 05:15:10 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 7 Dec 2009 18:45:10 +0530 Subject: [Freeswitch-users] Need Conference design help In-Reply-To: <26653473.post@talk.nabble.com> References: <26653473.post@talk.nabble.com> Message-ID: <7d79b3930912070515p8a5aae3tc8dc51f44dd6d80e@mail.gmail.com> Have a look at mod_conference http://wiki.freeswitch.org/wiki/Mod_conference On Sat, Dec 5, 2009 at 12:47 PM, shehzad p wrote: > > Hello Every one, > > I have to design conference, and I need community guidance to efficiently > accomplish that. > > I need to create Conference which will have three kind of users: > 1. Moderator (may be only one per conference) > 2. User who can participate in conference without moderator interaction. > 3. User who can only participate when Moderator allow them to get in. > > Also besides above setup I have to perform other things like Record the > conference, Multicast the conference to other freeswitch server. I saw the > conference Record CLI command but wondering where to setup when conference > starts. I am also wondering how Multicast Conference is possible in > Freeswitch and how the receiver Freeswitch configuration will look like. > > Thanks. > msp > > -- > View this message in context: > http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ba31c985/attachment-0002.html From erandr-junk at usa.net Mon Dec 7 07:28:52 2009 From: erandr-junk at usa.net (eaf) Date: Mon, 7 Dec 2009 07:28:52 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> Message-ID: <26678873.post@talk.nabble.com> Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently sitting in the first line of the top chart showing 3% CPU utilization when idle. Don't know whether it's because of the remaining epolls in sofia or whether it's because there are still some threads left in freeswitch that I neglected to change because they were sleeping with 100ms interval, so I figured, who cares. Maybe when all things come together (sofia, 100ms*N) freeswitch ends up spending 3% of CPU while doing pretty much nothing. Btw, compared with Asterisk, the latter is not even visible on the first top's screen and spends 1% CPU when bridging two G711 calls and recording them to disk. So, at this time I have both original Asterisk and FS setups running. One is seemless but clumsy in configuration, the other one is neat and stylish but too preoccupied with smth... Should I look into sofia epollers? That's kind of deep in the code. Or should I just stick with Asterisk? Anthony Minessale-2 wrote: > > There is another user here with a 300mhz box. I am willing to investigate > this improved performance for weak devices but I need to do it in a sane > cross-platform way. > > > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > > wrote: > >> A word to the wise to the general FreeSWITCH community: If Anthony >> Minessale suggests that you try to do any number of things, it's a very >> good idea to try all those ideas before continuing on. I've known him, >> MikeJ, and bkw for several years, and they almost always have very good >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely >> frustrating to try to help people out who won't try the provided >> suggestions first. >> >> And note directly to "eaf" - bogomips is quite possibly the least >> significant bit of data about a cpu that you will get out of >> /proc/cpuinfo... The name itself - bogo, means bogus. >> http://en.wikipedia.org/wiki/Bogomips >> >> -Yossi >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Mon Dec 7 07:44:24 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 7 Dec 2009 07:44:24 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <2FE67B50CF5C456E9958B83513618E3F@greyhawk.tonecommander.com> I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/08a565af/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 08:00:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 10:00:17 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26678873.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> Message-ID: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > Here is what I found... > > I tried high-priority scheduling as per your suggestion, reniced the > program > explicitly, rewrote timer thread to sleep on cond. variable and activate > only when there are timers and only when the timer actually had to be > clicked, turned off SQL thread and removed polling from sofia profile > thread. > > That pretty much eliminated all idle 1ms sleepers that were there except > for > three in sofia itself (su_epoll_port). And when I was about to be happy, I > found that two outgoing calls through my VOIP providers when bridged > together showed terrible distortions. I undid all my changes, tried 1.0.4, > trunk (noticed btw that when I bridge two calls via loopback in JS in the > trunk I must keep JS running, or the calls get terminated - NOT the same as > in 1.0.4 where exitting JS left calls running), got pretty much the same > sad > results. At the same time calls bridged by freeswitch between LAN and any > of > the VOIP providers behaved just fine. And calls bridged by Asterisk any way > were fine too. So that pretty much looked like the end of the freeswitch > trials for me. > > But then I timed your code, mine and found that all those 1ms sleeps that > your timer thread was doing (and all those pollers were doing as well) were > actually 4ms sleeps because you know what unless kernel is configured with > HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > (HZ=100). Mine was 250. > > This actually meant that the original timer thread was firing once, > sleeping > for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 > times back-to-back, etc. It was still firing 20ms timers on time, but 30ms > ones of course were not, since 30ms doesn't divide by 4 evenly. Plus > whoever > relied on runtime.reference or switch_micro_time_now() were kind of screwed > because both were running jumpy. Plus whoever assumed that apr_sleep(1000) > or cond_yield() was sleeping for 1ms were also in for a surprise. It felt > satisfying to find that, however it didn't explain why the same distortions > were observed with rewritten timer thread and disabled RTP timers. > > Anyway, I sighed (pretty much like you) and recompiled the kernel with > HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes > south, > you need to hook up serial console and see what the heck went wrong. > > That eliminated distortions, ha! But made freeswitch more CPU hungry. Now > the remaining 1ms threads sitting in sofia epoll were really polling for > 1ms, not 4, and freeswitch was consistently sitting in the first line of > the > top chart showing 3% CPU utilization when idle. > > Don't know whether it's because of the remaining epolls in sofia or whether > it's because there are still some threads left in freeswitch that I > neglected to change because they were sleeping with 100ms interval, so I > figured, who cares. Maybe when all things come together (sofia, 100ms*N) > freeswitch ends up spending 3% of CPU while doing pretty much nothing. > > Btw, compared with Asterisk, the latter is not even visible on the first > top's screen and spends 1% CPU when bridging two G711 calls and recording > them to disk. > > So, at this time I have both original Asterisk and FS setups running. One > is > seemless but clumsy in configuration, the other one is neat and stylish but > too preoccupied with smth... Should I look into sofia epollers? That's kind > of deep in the code. Or should I just stick with Asterisk? > > > > > > Anthony Minessale-2 wrote: > > > > There is another user here with a 300mhz box. I am willing to > investigate > > this improved performance for weak devices but I need to do it in a sane > > cross-platform way. > > > > > > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > > >> wrote: > > > >> A word to the wise to the general FreeSWITCH community: If Anthony > >> Minessale suggests that you try to do any number of things, it's a very > >> good idea to try all those ideas before continuing on. I've known him, > >> MikeJ, and bkw for several years, and they almost always have very good > >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > >> frustrating to try to help people out who won't try the provided > >> suggestions first. > >> > >> And note directly to "eaf" - bogomips is quite possibly the least > >> significant bit of data about a cpu that you will get out of > >> /proc/cpuinfo... The name itself - bogo, means bogus. > >> http://en.wikipedia.org/wiki/Bogomips > >> > >> -Yossi > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/a83d6006/attachment-0002.html From mike at jerris.com Mon Dec 7 08:16:30 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 7 Dec 2009 11:16:30 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: Also I have seen some people reporting that the new tickless timers in newer kernels work better. You may want to try those. Mike On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others work better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > Here is what I found... > > I tried high-priority scheduling as per your suggestion, reniced the program > explicitly, rewrote timer thread to sleep on cond. variable and activate > only when there are timers and only when the timer actually had to be > clicked, turned off SQL thread and removed polling from sofia profile > thread. > > That pretty much eliminated all idle 1ms sleepers that were there except for > three in sofia itself (su_epoll_port). And when I was about to be happy, I > found that two outgoing calls through my VOIP providers when bridged > together showed terrible distortions. I undid all my changes, tried 1.0.4, > trunk (noticed btw that when I bridge two calls via loopback in JS in the > trunk I must keep JS running, or the calls get terminated - NOT the same as > in 1.0.4 where exitting JS left calls running), got pretty much the same sad > results. At the same time calls bridged by freeswitch between LAN and any of > the VOIP providers behaved just fine. And calls bridged by Asterisk any way > were fine too. So that pretty much looked like the end of the freeswitch > trials for me. > > But then I timed your code, mine and found that all those 1ms sleeps that > your timer thread was doing (and all those pollers were doing as well) were > actually 4ms sleeps because you know what unless kernel is configured with > HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > (HZ=100). Mine was 250. > > This actually meant that the original timer thread was firing once, sleeping > for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 > times back-to-back, etc. It was still firing 20ms timers on time, but 30ms > ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever > relied on runtime.reference or switch_micro_time_now() were kind of screwed > because both were running jumpy. Plus whoever assumed that apr_sleep(1000) > or cond_yield() was sleeping for 1ms were also in for a surprise. It felt > satisfying to find that, however it didn't explain why the same distortions > were observed with rewritten timer thread and disabled RTP timers. > > Anyway, I sighed (pretty much like you) and recompiled the kernel with > HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, > you need to hook up serial console and see what the heck went wrong. > > That eliminated distortions, ha! But made freeswitch more CPU hungry. Now > the remaining 1ms threads sitting in sofia epoll were really polling for > 1ms, not 4, and freeswitch was consistently sitting in the first line of the > top chart showing 3% CPU utilization when idle. > > Don't know whether it's because of the remaining epolls in sofia or whether > it's because there are still some threads left in freeswitch that I > neglected to change because they were sleeping with 100ms interval, so I > figured, who cares. Maybe when all things come together (sofia, 100ms*N) > freeswitch ends up spending 3% of CPU while doing pretty much nothing. > > Btw, compared with Asterisk, the latter is not even visible on the first > top's screen and spends 1% CPU when bridging two G711 calls and recording > them to disk. > > So, at this time I have both original Asterisk and FS setups running. One is > seemless but clumsy in configuration, the other one is neat and stylish but > too preoccupied with smth... Should I look into sofia epollers? That's kind > of deep in the code. Or should I just stick with Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/31a8a586/attachment-0002.html From lfurrea at gmail.com Mon Dec 7 08:28:38 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Mon, 7 Dec 2009 10:28:38 -0600 Subject: [Freeswitch-users] Sporadic call drops In-Reply-To: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> References: <191c3a030912041151n45daedbh805495093b3fd777@mail.gmail.com> Message-ID: I will certainly shchedule time for the upgrade. Thanks for the answer On Fri, Dec 4, 2009 at 1:51 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > we changed that message a long time ago so people would not think that > anymore > We are now 3000 rev beyond the version you are at, I would like it if you > try the lastest trunk. > > > On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea wrote: > >> Hi all, >> >> Guys I know the question could be too vague, but I have a customer that >> just reported frequent failure to place outbound calls though a PSTN gateway >> on the LAN. >> >> I looked at the logs and I seem to be able to confirm that FS fails to >> place the call through the gateway and that the issue resides on the FS side >> since the first channel that s killed is tht of the internal extension >> registered to FS and then FS send the BYE to gw and kills the channel. >> >> What are possible causes of this? >> >> I know you always like to look at complete logs but here's a snip that >> could shed some light on the disconnection. (I can provide full logs if >> required and worthed) >> >> 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() >> Channel sofia/internal/200 at 172.16.3.5 entering state [ready][200] >> 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() >> Channel sofia/internal/200 at 172.16.3.5 entering state [terminated][200] >> 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() >> Hangup sofia/internal/200 at 172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal sofia/internal/200 at 172.16.3.5[KILL] >> 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal sofia/internal/ >> 200 at 172.16.3.5 [BREAK] >> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() >> sofia/internal/200 at 172.16.3.5 ending bridge by request from write >> function >> 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() >> sofia/pstn/22909980 at 172.16.3.46 receive message [UNBRIDGE] >> >> >> Is the 6th line normal behavior for ending the channel? >> >> FreeSWITCH Version 1.0.trunk (13484M) >> >> TIA >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4db8a923/attachment-0002.html From djbinter at yahoo.com Mon Dec 7 08:42:57 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 08:42:57 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <168319.49226.qm@web37502.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> Message-ID: <987536.45831.qm@web37508.mail.mud.yahoo.com> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. ________________________________ From: DJB To: FREESWITCH-USERS MAILING LIST Sent: Sun, December 6, 2009 5:17:14 PM Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -------------------------------------------------------------------------------------------------------------- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? S References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> Message-ID: <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > Any suggestion would be appreciated. Thank you. > > ------------------------------ > *From:* DJB > *To:* FREESWITCH-USERS MAILING LIST > > *Sent:* Sun, December 6, 2009 5:17:14 PM > *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with > high priority enabled. > > I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version > 1.0.4 (exported) with only one thing difference which is the first one is > running with -hp enabled; however, I have noticed that the one with -hp > option consumed double in memory usage than the other one. > > I wonder whether anyone can explain why. Thank you. > > Please see below: > > > -------------------------------------------------------------------------------------------------------------- > top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 > Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie > Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers > Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch > > 4418 session(s) 14/100 > > root 30750 2.1 *19.9* 1879252 1634300 ? S /usr/local/freeswitch/bin/freeswitch -nc -hp > > > -------------------------------------------------------------------------------------------------------------- > top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 > Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie > Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, > 0.0%st > Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers > Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch > > 4478 session(s) 14/100 > > root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 > /usr/local/freeswitch/bin/freeswitch -nc > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/66fc8548/attachment-0002.html From djbinter at yahoo.com Mon Dec 7 09:12:18 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 09:12:18 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> Message-ID: <794775.77398.qm@web37501.mail.mud.yahoo.com> Anthony, Thank you for your clear response. Based on your recommendation, if I want to route more calls to the first server, should I take off "-hp", or it's better to run with it. We are running FS for pass-thru traffic with signaling only. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Mon, December 7, 2009 8:56:14 AM Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. > > > ________________________________ From: DJB >To: FREESWITCH-USERS MAILING LIST >Sent: Sun, December 6, 2009 5:17:14 PM >Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. > > > >I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. > > >I wonder whether anyone can explain why. Thank you. > > >Please see below: > > >-------------------------------------------------------------------------------------------------------------- >top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 >Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, > 0 zombie >Cpu(s): 0.9%us, 0.2%sy, > 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st >Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers >Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch > > >4418 session(s) 14/100 > > >root 30750 2.1 19.9 1879252 1634300 > ? S > >-------------------------------------------------------------------------------------------------------------- >>top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 >Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st >Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers >Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached > > > PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND > > 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch > > >4478 session(s) 14/100 > > >root 7147 1.9 9.4 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc > > > > > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/46f0e3dc/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 09:31:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:31:27 -0600 Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <794775.77398.qm@web37501.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <191c3a030912070856r3e0783a4xb4613d0cee082596@mail.gmail.com> <794775.77398.qm@web37501.mail.mud.yahoo.com> Message-ID: <191c3a030912070931j5c96576fsede73fd56bedeca0@mail.gmail.com> maybe you can try both ways and see if there is a significant difference? I think -hp would help more if you were doing media than if you were not but that does not mean it could not still help performance but really the extra performance would only show up once you had consumed all the resources the box had to offer without -hp enabled in most cases. On Mon, Dec 7, 2009 at 11:12 AM, DJB wrote: > Anthony, > > Thank you for your clear response. Based on your recommendation, if I want > to route more calls to the first server, should I take off "-hp", or it's > better to run with it. We are running FS for pass-thru traffic with > signaling only. > > > ------------------------------ > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Mon, December 7, 2009 8:56:14 AM > *Subject:* Re: [Freeswitch-users] Question regarding running FreeSWITCH > with high priority enabled. > > One of the properties of -hp is to enable memlockall() which means disable > swapping. This causes all memory used by FS to be resident permanently and > is much more costly in memory usage. -hp also uses a RR scheduler runs the > process at a less nice level and increases a few other process ulimits. > This mode is designed for high end usage and uses more resources when idle > with a large payout when scaling to many calls. > > > > > On Mon, Dec 7, 2009 at 10:42 AM, DJB wrote: > >> One thing that I forgot to mention, these 2 FreeSWITCH servers are getting >> calls with load balancing from another switch. Thus, the traffic type are >> pretty much identical and both FSs have exactly the same on configuration. >> Any suggestion would be appreciated. Thank you. >> >> ------------------------------ >> *From:* DJB >> *To:* FREESWITCH-USERS MAILING LIST < >> freeswitch-users at lists.freeswitch.org> >> *Sent:* Sun, December 6, 2009 5:17:14 PM >> *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with >> high priority enabled. >> >> I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version >> 1.0.4 (exported) with only one thing difference which is the first one is >> running with -hp enabled; however, I have noticed that the one with -hp >> option consumed double in memory usage than the other one. >> >> I wonder whether anyone can explain why. Thank you. >> >> Please see below: >> >> >> -------------------------------------------------------------------------------------------------------------- >> top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 >> Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >> Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, >> 0.0%st >> Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers >> Swap: 10223608k total, 0k used, 10223608k free, 5417524k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch >> >> 4418 session(s) 14/100 >> >> root 30750 2.1 *19.9* 1879252 1634300 ? S> /usr/local/freeswitch/bin/freeswitch -nc -hp >> >> >> -------------------------------------------------------------------------------------------------------------- >> top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 >> Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie >> Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, >> 0.0%st >> Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers >> Swap: 10223608k total, 0k used, 10223608k free, 5432632k cached >> >> PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND >> >> 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch >> >> >> 4478 session(s) 14/100 >> >> root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 >> /usr/local/freeswitch/bin/freeswitch -nc >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/e05eca83/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 09:32:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:32:25 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: References: <26594250.post@talk.nabble.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <191c3a030912070932m1c17b98dnd09181dcfe53d1f1@mail.gmail.com> oh and also use top -H to see which threads are using specific CPU and try to cross reference them by attaching with gdb and dumping all the thread bt On Mon, Dec 7, 2009 at 10:16 AM, Michael Jerris wrote: > Also I have seen some people reporting that the new tickless timers in > newer kernels work better. You may want to try those. > > Mike > > On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: > > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm. We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using. I am curious > what would happen if you install Kristian's astlinux on one of your devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do > these systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread > which will make all the switch_cond_next share a 1ms conditional instead of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others > work better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > >> >> Here is what I found... >> >> I tried high-priority scheduling as per your suggestion, reniced the >> program >> explicitly, rewrote timer thread to sleep on cond. variable and activate >> only when there are timers and only when the timer actually had to be >> clicked, turned off SQL thread and removed polling from sofia profile >> thread. >> >> That pretty much eliminated all idle 1ms sleepers that were there except >> for >> three in sofia itself (su_epoll_port). And when I was about to be happy, I >> found that two outgoing calls through my VOIP providers when bridged >> together showed terrible distortions. I undid all my changes, tried 1.0.4, >> trunk (noticed btw that when I bridge two calls via loopback in JS in the >> trunk I must keep JS running, or the calls get terminated - NOT the same >> as >> in 1.0.4 where exitting JS left calls running), got pretty much the same >> sad >> results. At the same time calls bridged by freeswitch between LAN and any >> of >> the VOIP providers behaved just fine. And calls bridged by Asterisk any >> way >> were fine too. So that pretty much looked like the end of the freeswitch >> trials for me. >> >> But then I timed your code, mine and found that all those 1ms sleeps that >> your timer thread was doing (and all those pollers were doing as well) >> were >> actually 4ms sleeps because you know what unless kernel is configured with >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms >> (HZ=100). Mine was 250. >> >> This actually meant that the original timer thread was firing once, >> sleeping >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 >> times back-to-back, etc. It was still firing 20ms timers on time, but 30ms >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus >> whoever >> relied on runtime.reference or switch_micro_time_now() were kind of >> screwed >> because both were running jumpy. Plus whoever assumed that apr_sleep(1000) >> or cond_yield() was sleeping for 1ms were also in for a surprise. It felt >> satisfying to find that, however it didn't explain why the same >> distortions >> were observed with rewritten timer thread and disabled RTP timers. >> >> Anyway, I sighed (pretty much like you) and recompiled the kernel with >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes >> south, >> you need to hook up serial console and see what the heck went wrong. >> >> That eliminated distortions, ha! But made freeswitch more CPU hungry. Now >> the remaining 1ms threads sitting in sofia epoll were really polling for >> 1ms, not 4, and freeswitch was consistently sitting in the first line of >> the >> top chart showing 3% CPU utilization when idle. >> >> Don't know whether it's because of the remaining epolls in sofia or >> whether >> it's because there are still some threads left in freeswitch that I >> neglected to change because they were sleeping with 100ms interval, so I >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. >> >> Btw, compared with Asterisk, the latter is not even visible on the first >> top's screen and spends 1% CPU when bridging two G711 calls and recording >> them to disk. >> >> So, at this time I have both original Asterisk and FS setups running. One >> is >> seemless but clumsy in configuration, the other one is neat and stylish >> but >> too preoccupied with smth... Should I look into sofia epollers? That's >> kind >> of deep in the code. Or should I just stick with Asterisk? >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/89804735/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 09:35:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 11:35:04 -0600 Subject: [Freeswitch-users] lua+sqlite example? In-Reply-To: <04a201ca7623$2c0b2020$84216060$@com> References: <04a201ca7623$2c0b2020$84216060$@com> Message-ID: <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein wrote: > Greetings. We are attempting to add sqlite access to an IVR application > we are prototyping. We are using lua for the scripts. Is there an example > anywhere of a lua + sqlite script? Do we need to install luasql? Any > help/pointers greatly appreciated. > > > > --Steve Klein > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d5be5e76/attachment-0002.html From msc at freeswitch.org Mon Dec 7 09:43:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 09:43:17 -0800 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <4B1CAF25.6010706@greatiam.com> References: <4B1CAF25.6010706@greatiam.com> Message-ID: <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah wrote: > Pardon me if this has been addressed already. > How does one go about having in the simplest instance 2 servers > registering with each other on startup whereby the users registering > would be able to call each other. > The 2 servers are in different domains. > > Thanks. > Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/c2605e5c/attachment-0002.html From ken at ksac.com Mon Dec 7 09:27:49 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 09:27:49 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 Message-ID: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/2db542cd/attachment-0002.html From brian at freeswitch.org Mon Dec 7 10:10:11 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 12:10:11 -0600 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get > esl to compile. My version is earlier than the snow leopard that is > mentioned in the general install docs, and I have tried it with and > without the compiler flags in the freewswtch installation -> MAC os X. > I have also googled this, and don?t see what I am doing wrong. > Anybody there that can help? > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/ > freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../ > libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable - > Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="- > I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g - > ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc - > lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > _main > __convert_to_string > __efree > __emalloc > __estrndup > __zend_get_parameters_array_ex > __zend_list_find > __zval_copy_ctor > _compiler_globals > _convert_to_long > _zend_error > _zend_get_constant > _zend_hash_find > _zend_register_list_destructors_ex > _zend_register_long_constant > _zend_register_resource > _zend_rsrc_list_get_rsrc_type > _zend_wrong_param_count > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make: *** [phpmod] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d9a2f13a/attachment-0002.html From mcampbellsmith at gmail.com Mon Dec 7 10:11:17 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 05:11:17 +1100 Subject: [Freeswitch-users] Access to users variables Message-ID: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> Hi! How can I access the variables that are defined in a users xml file? For example, say user 1000 has a variable called smsnumber, as defined below: How can I use variable ${smsnumber} in a dialplan to run a perl script using ? Thanks From msc at freeswitch.org Mon Dec 7 10:21:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:21:36 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> Forgive me if I ask the obvious questions... Did you "make" in src/libs/esl before doing "make phpmod" ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer wrote: > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get esl to > compile. My version is earlier than the snow leopard that is mentioned in > the general install docs, and I have tried it with and without the compiler > flags in the freewswtch installation -> MAC os X. > > I have also googled this, and don?t see what I am doing wrong. Anybody > there that can help? > > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" > CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE > -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc > -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > > _main > > __convert_to_string > > __efree > > __emalloc > > __estrndup > > __zend_get_parameters_array_ex > > __zend_list_find > > __zval_copy_ctor > > _compiler_globals > > _convert_to_long > > _zend_error > > _zend_get_constant > > _zend_hash_find > > _zend_register_list_destructors_ex > > _zend_register_long_constant > > _zend_register_resource > > _zend_rsrc_list_get_rsrc_type > > _zend_wrong_param_count > > collect2: ld returned 1 exit status > > make[1]: *** [ESL.so] Error 1 > > make: *** [phpmod] Error 2 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/6825e5d6/attachment-0002.html From msc at freeswitch.org Mon Dec 7 10:25:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:25:36 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> Message-ID: <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > How can I access the variables that are defined in a users xml file? > > For example, say user 1000 has a variable called smsnumber, as defined > below: > > > > > > > > > > > > > How can I use variable ${smsnumber} in a dialplan to run a perl script > using ? > > Do you just want to pass the value in smsnumber to the sms.pl script? Have you tried this? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/b82afa81/attachment-0002.html From Prometheus001 at gmx.net Mon Dec 7 10:31:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 07 Dec 2009 19:31:36 +0100 Subject: [Freeswitch-users] continue_on_fail Message-ID: <4B1D4A08.80507@gmx.net> I have a Problem with continue_on_fail. I have setup a hunt group and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter From ken at ksac.com Mon Dec 7 10:33:44 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 10:33:44 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> References: <87f2f3b90912071021x6ccc9e3rf40a10ab82284a02@mail.gmail.com> Message-ID: I did make first, but did not install any extra dev stuff, thinking I already had them. Is there a way to turn on verbose and finding out exactly what it no there that is expected? Thanksmuch!! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, December 07, 2009 1:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 Forgive me if I ask the obvious questions... Did you "make" in src/libs/esl before doing "make phpmod" ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer > wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/d476470b/attachment-0002.html From ken at ksac.com Mon Dec 7 10:34:41 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 7 Dec 2009 10:34:41 -0800 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, December 07, 2009 1:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/e0470b04/attachment-0002.html From mcampbellsmith at gmail.com Mon Dec 7 10:37:03 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 05:37:03 +1100 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> Message-ID: <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> Hi! That's exactly what I want to do and that was the first thing I tried, but nothing is passed to the script. In a case like this, what defines if variable smsnumber is taken from the A path or B path? (The A path does not have smsnumber defined) On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins wrote: > > > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> How can I access the variables that are defined in a users xml file? >> >> For example, say user 1000 has a variable called smsnumber, as defined >> below: >> >> >> ? >> ? ? >> ? ? ? >> ? ? >> ? ? >> ? ? ? >> ? ? >> ? >> >> >> How can I use variable ${smsnumber} in a dialplan to run a perl script >> using ? >> > > Do you just want to pass the value in smsnumber to the sms.pl script? Have > you tried this? > > > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jmesquita at freeswitch.org Mon Dec 7 10:44:34 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 7 Dec 2009 16:44:34 -0200 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> Message-ID: Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > I installed libjpeg-7 following this website: > http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And > the previous error is replaced by a new one: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. > -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 > -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes > -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF > .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o > at_interpreter.o > at_interpreter.c: In function ???command_search???: > at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use > in this function) > at_interpreter.c:5299: error: (Each undeclared identifier is reported only > once > at_interpreter.c:5299: error: for each function it appears in.) > at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in > this function) > at_interpreter.c: In function ???at_interpreter???: > at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in > this function) > make[8]: *** [at_interpreter.lo] Error 1 > > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > However, I'm still able to start freeswitch and mod_skypiax and make skype > calls with no problem. > > Regards, > -Jingwei > > > > On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > >> No, I didn't change or update the system libs. I just wanted to double >> check whether my system has this libjpeg library. ./configure was definitely >> executed before the source codes were rebuilt. >> >> Regards, >> -Jingwei >> >> >> On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: >> >>> Hi, >>> >>> That one is on your side. If you changed/updated system libs it might be >>> worth doing another ./configure >>> >>> Cheers, >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >>> >>> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >>> However, I encounter another one. >>> >>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>> -lc >>> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: >>> cannot open shared object file: No such file or directory >>> make[8]: *** [at_interpreter_dictionary.h] Error 127 >>> make[7]: *** [all] Error 2 >>> make[6]: *** [all-recursive] Error 1 >>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_voipcodecs-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> Do you have idea about this one? >>> >>> Thanks! >>> >>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >>> >>>> Consider it fixed. >>>> Committed revision 15765. >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>>> >>>> Hi Guys, >>>> >>>> I got a compilation error of skypiax_protocol.c with the latest version >>>> r15764. >>>> >>>> Compiling skypiax_protocol.c... >>>> *cc1: warnings being treated as errors* >>>> skypiax_protocol.c: In function ???X11_errors_handler???: >>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c: In function ???skypiax_send_message???: >>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and >>>> code >>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and >>>> code >>>> make[5]: *** [skypiax_protocol.o] Error 1 >>>> make[4]: *** [install] Error 1 >>>> make[3]: *** [mod_skypiax-install] Error 1 >>>> make[2]: *** [install-recursive] Error 1 >>>> >>>> I personally checked the file and it shouldn't be a merge problem. Does >>>> anyone encounter this as well? >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4183f10a/attachment-0002.html From jerry.richards at teotech.com Mon Dec 7 10:49:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 7 Dec 2009 10:49:01 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/73e6eaec/attachment-0002.html From abeka at greatiam.com Mon Dec 7 10:50:58 2009 From: abeka at greatiam.com (Otis) Date: Mon, 07 Dec 2009 18:50:58 +0000 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> References: <4B1CAF25.6010706@greatiam.com> <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> Message-ID: <4B1D4E92.1040204@greatiam.com> Michael Collins wrote: > > > On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah > > wrote: > > Pardon me if this has been addressed already. > How does one go about having in the simplest instance 2 servers > registering with each other on startup whereby the users registering > would be able to call each other. > The 2 servers are in different domains. > > Thanks. > > > Are the two servers in different locations? Different LANs? Is NAT > involved? Just checking. Really this is just a matter of loading the > default config on each machine and then making some decisions about > the dialplan: do you want prefix dialing so that you can have ext 1000 > at both locations or do you want to have something like 1000~1099 at > location A and 1100~1199 at location B? From there it's just a matter > of creating the gateways on each machine and adding a dialplan entry > to handle the routing. > -MC > Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC Thanks . From msc at freeswitch.org Mon Dec 7 10:56:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 10:56:21 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> Message-ID: <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user you might just need to set the user so that the vars become available on the leg you're processing. -MC On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > That's exactly what I want to do and that was the first thing I tried, > but nothing is passed to the script. > > In a case like this, what defines if variable smsnumber is taken from > the A path or B path? (The A path does not have smsnumber defined) > > On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins > wrote: > > > > > > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith > > wrote: > >> > >> Hi! > >> > >> How can I access the variables that are defined in a users xml file? > >> > >> For example, say user 1000 has a variable called smsnumber, as defined > >> below: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> How can I use variable ${smsnumber} in a dialplan to run a perl script > >> using ? > >> > > > > Do you just want to pass the value in smsnumber to the sms.pl script? > Have > > you tried this? > > > > > > > > -MC > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/53615f4b/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 11:01:59 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 19:01:59 -0000 Subject: [Freeswitch-users] no hangup on B leg Message-ID: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/9c947a67/attachment-0002.html From mcampbellsmith at gmail.com Mon Dec 7 11:09:55 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 8 Dec 2009 06:09:55 +1100 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> Message-ID: <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> Perfect... works like a charm. Thanks Mike. On Tue, Dec 8, 2009 at 5:56 AM, Michael Collins wrote: > Check out > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user > > you might just need to set the user so that the vars become available on the > leg you're processing. > -MC > > On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> That's exactly what I want to do and that was the first thing I tried, >> but nothing is passed to the script. >> >> In a case like this, what defines if variable smsnumber is taken from >> the A path or B path? (The A path does not have smsnumber defined) >> >> On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins >> wrote: >> > >> > >> > On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith >> > wrote: >> >> >> >> Hi! >> >> >> >> How can I access the variables that are defined in a users xml file? >> >> >> >> For example, say user 1000 has a variable called smsnumber, as defined >> >> below: >> >> >> >> >> >> ? >> >> ? ? >> >> ? ? ? >> >> ? ? >> >> ? ? >> >> ? ? ? >> >> ? ? >> >> ? >> >> >> >> >> >> How can I use variable ${smsnumber} in a dialplan to run a perl script >> >> using ? >> >> >> > >> > Do you just want to pass the value in smsnumber to the sms.pl script? >> > Have >> > you tried this? >> > >> > >> > >> > -MC >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Dec 7 11:11:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 11:11:36 -0800 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: References: Message-ID: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi all, > > > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging > up both legs on a bridge, now today, with nothing changed, I?m not seeing a > hangup of the b leg at all. > > > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls > hangup just fine. Before when I had an issue with the B leg not closing the > bridge, I was at least getting a hangup event, now it?s not being fired. > Does anyone have an idea what might be causing this? > > > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/6ab3e27e/attachment-0002.html From Prometheus001 at gmx.net Mon Dec 7 11:18:27 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 07 Dec 2009 20:18:27 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B1AC410.9050201@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> <4B1AC410.9050201@gmx.net> Message-ID: <4B1D5503.8010308@gmx.net> Hello, i now changed the $${domain} name of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Best regards Peter Peter P GMX schrieb: > Hello Anthony, > > I did some checks today > Here is how the phones are registered: > > mysql> select sip_host, presence_hosts, server_user,server_host, > hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip_host | presence_hosts | server_user | server_host | > hostname | sip_realm | mwi_user | mwi_host | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | > sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > IPs are: > 10.11.12.1 sip1.mydomain.com (common cluster IP) > 10.11.12.2 sip11.mydomain.com > 10.11.12.3 sip12.mydomain.com (not used at this point) > > XML-Curl for the directory is: > >
> > > > > > > > > value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > >
> > > > The internal profile has the following alias: > > > > > > > With $${domain} being sip11.mydomain.com > > Phones are registering to sip1.mydomain.com, Voicemail works, but MWI > does not. Any hint what I should change to make this work? > > Best regards > Peter > > Anthony Minessale schrieb: > >> based on your example past >> >> sip1.mydomain.com is the domain in the >> packet and thus the profile should have an alias for this. >> Then the user must reside in your sip db with the user 200 and domain >> sip1.mydomain.com >> >> if you dont have this consider the force-register-domain and >> force-register-db-domain to normalize the host names. >> >> >> On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale >> > wrote: >> >> Did you check the 2 replies that told you you need aliases in your >> sofia profile to translate the domain found in your >> message_waiting to the right profile? Both Brian and Mike >> answered you. >> >> >> >> >> >> On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX >> > wrote: >> >> I tried now with phones directly attached to the freeswitch >> (without an >> OpenSIPS in between). I also added the alias. But the >> behaviour is as >> before: >> No notify message from freeswitch, neither after register nor >> after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on >> the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > From anthony.minessale at gmail.com Mon Dec 7 11:39:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 13:39:52 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> try rerunning the ./bootstrap.sh On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards wrote: > When I got the latest trunk the make gets an error. Should I perhaps > disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. > Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 7:44 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs > without SDP. I will try to update to the latest trunk today and capture > traces as Anthony described. If I can't do it today, it might be at the end > of the week. > > Best Regards, > Jerry > > > ------------------------------ > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Saturday, December 05, 2009 7:30 PM > *To:* Jerry Richards > *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > > Why are you changing the 3pcc setting, is this an invite with no sdp? > you need to take a trace from FS. > > 1) update to latest trunk first so line number match up. > 2) issue these commands > > sofia profile internal siptrace on > console loglevel debug > > save the output and put it on pastebin http://pastebin.freeswitch.org > > > > > On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards > wrote: > >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/2e81a903/attachment-0002.html From msc at freeswitch.org Mon Dec 7 11:45:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 11:45:05 -0800 Subject: [Freeswitch-users] Access to users variables In-Reply-To: <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> References: <33c87fa30912071011vf869e0duf1c9b5cca853f7a3@mail.gmail.com> <87f2f3b90912071025w55a42c4dp25460161d9c1af08@mail.gmail.com> <33c87fa30912071037g33956494g71d06649353f46f1@mail.gmail.com> <87f2f3b90912071056w37eafd74l34e2d0257aad29d9@mail.gmail.com> <33c87fa30912071109m65e6aea2sd3ebd3fd9f4b03a5@mail.gmail.com> Message-ID: <87f2f3b90912071145o2bb416fasfcd278c766347f47@mail.gmail.com> On Mon, Dec 7, 2009 at 11:09 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Perfect... > > > > works like a charm. > "Another satisfied customer!" :P -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ac321d49/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 11:59:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 13:59:13 -0600 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <191c3a030912071159o140740b7y7214ac022cfdffc2@mail.gmail.com> also, don't use 1.0.4, please us the latest SVN or last svn snapshot at the very least. On Mon, Dec 7, 2009 at 12:34 PM, Kendall Stauffer wrote: > Any direction on where to start would be appreciated. I am trying to get > freepbx working with this, and everything works (I think) except esl > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Monday, December 07, 2009 1:10 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] esl for Mac OS X 10.4 > > > > The build system for libesl and everything below that won't work 100% on > the mac just yet. You have to make some changes to how its linked and > you'll have to compile php yourself to get everything in there properly. > The perl one however is much easier to fix. > > > > -SOLINK=-shared -Xlinker -x > > +SOLINK=-dynamiclib -Xlinker -x > > > > > > Thats all you usually fix for the mac. > > > > > > /b > > > > > > > > On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > > > > I have downloaded and compiled freeswitch, and it runs fine, can > compile everything without error including spandsp, but can?t get esl to > compile. My version is earlier than the snow leopard that is mentioned in > the general install docs, and I have tried it with and without the compiler > flags in the freewswtch installation -> MAC os X. > > I have also googled this, and don?t see what I am doing wrong. Anybody > there that can help? > > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make > phpmod-install > > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror > -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" > CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE > -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" > CXX_CFLAGS="" -C php > > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc > -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > > _main > > __convert_to_string > > __efree > > __emalloc > > __estrndup > > __zend_get_parameters_array_ex > > __zend_list_find > > __zval_copy_ctor > > _compiler_globals > > _convert_to_long > > _zend_error > > _zend_get_constant > > _zend_hash_find > > _zend_register_list_destructors_ex > > _zend_register_long_constant > > _zend_register_resource > > _zend_rsrc_list_get_rsrc_type > > _zend_wrong_param_count > > collect2: ld returned 1 exit status > > make[1]: *** [ESL.so] Error 1 > > make: *** [phpmod] Error 2 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/dd9a5c4b/attachment-0002.html From sklein at singular.com Mon Dec 7 12:19:01 2009 From: sklein at singular.com (Steve Klein) Date: Mon, 7 Dec 2009 12:19:01 -0800 Subject: [Freeswitch-users] Database suggestions/pointers/? In-Reply-To: <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> References: <053d01ca76af$c311d2c0$49357840$@com> <855e4dcf0912061659k795fab82i7b28d318c4e30440@mail.gmail.com> Message-ID: <06c901ca777a$845ec800$8d1c5800$@com> Thanks for the suggestions. We'll explore. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Uckun Sent: Sunday, December 06, 2009 5:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Database suggestions/pointers/? On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein wrote: > Greetings. We need to add database access to an IVR application we are > prototyping. Based on FS best practice suggestions, we are using Lua for > the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite > would be a recommended approach. However, we cant find any examples of this > combo anywhere. So, what is the best practice scripting + database > recommendation for a high-volume database-driven FS app? > I would suggest you take a look at freeswitcher (http://github.com/bougyman/freeswitcher). The good thing is that it's ruby and therefore you can use any database compatible with ruby (that's all of them pretty much). You can also use an ORM of your choice or if you don't want to use an ORM you can use the amazingly fantastic sequel library. Being ruby it will run outside of the freeswitch memory space and you will have to use the inbound/outbound socket API. That may be a good thing if you want to separate your database and IVR logic from the machine running your freeswitch. Ruby is pretty easy to pick up if you don't know it and there are a wealth of libraries if you want to do other things like connect to web sites, manipulate XML, etc. There is also a liverpie http://github.com/jsgoecke/liverpie which is more of a proxy thing you can interface with any language. I am sure lua is nice but it seems like people are having some problems with ODBC, memory leaks etc when it comes to databases. If you go a ruby library that all goes away. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sklein at singular.com Mon Dec 7 12:19:48 2009 From: sklein at singular.com (Steve Klein) Date: Mon, 7 Dec 2009 12:19:48 -0800 Subject: [Freeswitch-users] lua+sqlite example? In-Reply-To: <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> References: <04a201ca7623$2c0b2020$84216060$@com> <191c3a030912070935u183ff728j8b2c99576da1f5b8@mail.gmail.com> Message-ID: <06ca01ca777a$a04125e0$e0c371a0$@com> Thanks. We'll look at that. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, December 07, 2009 9:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] lua+sqlite example? yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein wrote: Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.426 / Virus Database: 270.14.83/2529 - Release Date: 12/07/09 07:33:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ec119444/attachment-0002.html From mailinglist at fribert.dk Mon Dec 7 12:50:39 2009 From: mailinglist at fribert.dk (mailinglist) Date: Mon, 07 Dec 2009 21:50:39 +0100 Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through Message-ID: <4B1D78AF020000E1000002BC@mail.fribert.dk> Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_sofia at 10.11.12.25:5060 BIND-URL sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1001 at 10.11.12.25 Contact: "1001" Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID: OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1002 at 10.11.12.25 Contact: "1002" Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 ================================================================================================= And... sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_sofia at 87.61.18.196:5080 BIND-URL sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 Registrations: ================================================================================================= ================================================================================================= In my Dialplan I've created these two entries: ----- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- and ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$1 I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. If I try and make a local call from 1001 to 1002 it says 2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001 at 1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0d73912f/attachment-0002.html From erandr-junk at usa.net Mon Dec 7 12:58:55 2009 From: erandr-junk at usa.net (eaf) Date: Mon, 7 Dec 2009 12:58:55 -0800 (PST) Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <372NLBDsn2832S19.1259723979@cmsweb19.cms.usa.net> <191c3a030912021032j2a0639f3jde17e7d995a9a0f7@mail.gmail.com> <26619085.post@talk.nabble.com> <14CBC790-EDC1-49F2-9FC7-23F660666EE5@jerris.com> <26621005.post@talk.nabble.com> <26626634.post@talk.nabble.com> <26627246.post@talk.nabble.com> <191c3a030912030816w203d6e7cwdfb339ec74985fcc@mail.gmail.com> <26629856.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <26684048.post@talk.nabble.com> What do you want me to check while running these tests? Sound quality (it's good now even with original 1.0.4). Or CPU utilization? It's Debian 4. Anthony Minessale-2 wrote: > > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least > as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly > can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm. We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using. I am > curious > what would happen if you install Kristian's astlinux on one of your > devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do > these > systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer > thread > which will make all the switch_cond_next share a 1ms conditional instead > of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others > work > better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > >> >> Here is what I found... >> >> I tried high-priority scheduling as per your suggestion, reniced the >> program >> explicitly, rewrote timer thread to sleep on cond. variable and activate >> only when there are timers and only when the timer actually had to be >> clicked, turned off SQL thread and removed polling from sofia profile >> thread. >> >> That pretty much eliminated all idle 1ms sleepers that were there except >> for >> three in sofia itself (su_epoll_port). And when I was about to be happy, >> I >> found that two outgoing calls through my VOIP providers when bridged >> together showed terrible distortions. I undid all my changes, tried >> 1.0.4, >> trunk (noticed btw that when I bridge two calls via loopback in JS in the >> trunk I must keep JS running, or the calls get terminated - NOT the same >> as >> in 1.0.4 where exitting JS left calls running), got pretty much the same >> sad >> results. At the same time calls bridged by freeswitch between LAN and any >> of >> the VOIP providers behaved just fine. And calls bridged by Asterisk any >> way >> were fine too. So that pretty much looked like the end of the freeswitch >> trials for me. >> >> But then I timed your code, mine and found that all those 1ms sleeps that >> your timer thread was doing (and all those pollers were doing as well) >> were >> actually 4ms sleeps because you know what unless kernel is configured >> with >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms >> (HZ=100). Mine was 250. >> >> This actually meant that the original timer thread was firing once, >> sleeping >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing >> 4 >> times back-to-back, etc. It was still firing 20ms timers on time, but >> 30ms >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus >> whoever >> relied on runtime.reference or switch_micro_time_now() were kind of >> screwed >> because both were running jumpy. Plus whoever assumed that >> apr_sleep(1000) >> or cond_yield() was sleeping for 1ms were also in for a surprise. It felt >> satisfying to find that, however it didn't explain why the same >> distortions >> were observed with rewritten timer thread and disabled RTP timers. >> >> Anyway, I sighed (pretty much like you) and recompiled the kernel with >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes >> south, >> you need to hook up serial console and see what the heck went wrong. >> >> That eliminated distortions, ha! But made freeswitch more CPU hungry. Now >> the remaining 1ms threads sitting in sofia epoll were really polling for >> 1ms, not 4, and freeswitch was consistently sitting in the first line of >> the >> top chart showing 3% CPU utilization when idle. >> >> Don't know whether it's because of the remaining epolls in sofia or >> whether >> it's because there are still some threads left in freeswitch that I >> neglected to change because they were sleeping with 100ms interval, so I >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. >> >> Btw, compared with Asterisk, the latter is not even visible on the first >> top's screen and spends 1% CPU when bridging two G711 calls and recording >> them to disk. >> >> So, at this time I have both original Asterisk and FS setups running. One >> is >> seemless but clumsy in configuration, the other one is neat and stylish >> but >> too preoccupied with smth... Should I look into sofia epollers? That's >> kind >> of deep in the code. Or should I just stick with Asterisk? >> >> >> >> >> >> Anthony Minessale-2 wrote: >> > >> > There is another user here with a 300mhz box. I am willing to >> investigate >> > this improved performance for weak devices but I need to do it in a >> sane >> > cross-platform way. >> > >> > >> > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman >> > > >> wrote: >> > >> >> A word to the wise to the general FreeSWITCH community: If Anthony >> >> Minessale suggests that you try to do any number of things, it's a >> very >> >> good idea to try all those ideas before continuing on. I've known >> him, >> >> MikeJ, and bkw for several years, and they almost always have very >> good >> >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely >> >> frustrating to try to help people out who won't try the provided >> >> suggestions first. >> >> >> >> And note directly to "eaf" - bogomips is quite possibly the least >> >> significant bit of data about a cpu that you will get out of >> >> /proc/cpuinfo... The name itself - bogo, means bogus. >> >> http://en.wikipedia.org/wiki/Bogomips >> >> >> >> -Yossi >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26684048.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Dec 7 13:06:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 21:06:29 -0000 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/8df13f49/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 13:29:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 15:29:15 -0600 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <26684048.post@talk.nabble.com> References: <26594250.post@talk.nabble.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> <26684048.post@talk.nabble.com> Message-ID: <191c3a030912071329h374a1efcl240ae6e3e45e7810@mail.gmail.com> Both, if it always sounds ok then I guess CPU usage. On Mon, Dec 7, 2009 at 2:58 PM, eaf wrote: > > What do you want me to check while running these tests? Sound quality (it's > good now even with original 1.0.4). Or CPU utilization? > > It's Debian 4. > > > Anthony Minessale-2 wrote: > > > > Did you do each thing alone too to tell the difference? > > -hp alone, disable monotonic alone (i did not see you mention the disable > > monotonic) > > > > as for your 4ms thing, yes we require high resolution timing, if we ask > to > > sleep 1000 microseconds that is what we need it to sleep for or at least > > as > > close as possible, and the main reason that thread is never sleeping is > > because you can't actually count on it to run every 1ms but you mostly > > can. > > Hence the whole philosophy on only making 1 thread run hot all the time > to > > ensure that the rest don't have to repeat the same algorithm. We focus > on > > high end performance this was the point of your experimentation because > we > > will need to use a compile time defines and other logic to make it more > > efficient on your platform, a platform which we are not using. I am > > curious > > what would happen if you install Kristian's astlinux on one of your > > devices, > > i think you should also compare the kernel versions. > > > > > > What OS are you running anyway? > > > > Here are some more things to try (running plain trunk with no mods) do > > these > > systematically each alone and all together with/without -hp or disable > > monotonic etc to see what different combos create > > > > comment out this line (line 10) > > #define DISABLE_1MS_COND > > > > rebuild, this tells it to run a conditional at 1ms in the same timer > > thread > > which will make all the switch_cond_next share a 1ms conditional instead > > of > > doing microsleeps > > > > next > > > > some kernels/devices work better using select(0) for sleep where others > > work > > better using usleep. > > comment out line 109 > > apr_sleep(t); > > > > and try > > usleep(t) > > > > also mac works better using nanosleep so you could try changing it so it > > uses the code starting at 101 instead. > > > > > > also your claim about JS should be investigated because I do not think it > > should be the case. > > but you may want to move this to a jira http://jira.freeswitch.org > > > > As for the asterisk comparison, > > not sure how to answer you, that's your decision. > > > > > > > > On Mon, Dec 7, 2009 at 9:28 AM, eaf wrote: > > > >> > >> Here is what I found... > >> > >> I tried high-priority scheduling as per your suggestion, reniced the > >> program > >> explicitly, rewrote timer thread to sleep on cond. variable and activate > >> only when there are timers and only when the timer actually had to be > >> clicked, turned off SQL thread and removed polling from sofia profile > >> thread. > >> > >> That pretty much eliminated all idle 1ms sleepers that were there except > >> for > >> three in sofia itself (su_epoll_port). And when I was about to be happy, > >> I > >> found that two outgoing calls through my VOIP providers when bridged > >> together showed terrible distortions. I undid all my changes, tried > >> 1.0.4, > >> trunk (noticed btw that when I bridge two calls via loopback in JS in > the > >> trunk I must keep JS running, or the calls get terminated - NOT the same > >> as > >> in 1.0.4 where exitting JS left calls running), got pretty much the same > >> sad > >> results. At the same time calls bridged by freeswitch between LAN and > any > >> of > >> the VOIP providers behaved just fine. And calls bridged by Asterisk any > >> way > >> were fine too. So that pretty much looked like the end of the freeswitch > >> trials for me. > >> > >> But then I timed your code, mine and found that all those 1ms sleeps > that > >> your timer thread was doing (and all those pollers were doing as well) > >> were > >> actually 4ms sleeps because you know what unless kernel is configured > >> with > >> HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms > >> (HZ=100). Mine was 250. > >> > >> This actually meant that the original timer thread was firing once, > >> sleeping > >> for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing > >> 4 > >> times back-to-back, etc. It was still firing 20ms timers on time, but > >> 30ms > >> ones of course were not, since 30ms doesn't divide by 4 evenly. Plus > >> whoever > >> relied on runtime.reference or switch_micro_time_now() were kind of > >> screwed > >> because both were running jumpy. Plus whoever assumed that > >> apr_sleep(1000) > >> or cond_yield() was sleeping for 1ms were also in for a surprise. It > felt > >> satisfying to find that, however it didn't explain why the same > >> distortions > >> were observed with rewritten timer thread and disabled RTP timers. > >> > >> Anyway, I sighed (pretty much like you) and recompiled the kernel with > >> HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes > >> south, > >> you need to hook up serial console and see what the heck went wrong. > >> > >> That eliminated distortions, ha! But made freeswitch more CPU hungry. > Now > >> the remaining 1ms threads sitting in sofia epoll were really polling for > >> 1ms, not 4, and freeswitch was consistently sitting in the first line of > >> the > >> top chart showing 3% CPU utilization when idle. > >> > >> Don't know whether it's because of the remaining epolls in sofia or > >> whether > >> it's because there are still some threads left in freeswitch that I > >> neglected to change because they were sleeping with 100ms interval, so I > >> figured, who cares. Maybe when all things come together (sofia, 100ms*N) > >> freeswitch ends up spending 3% of CPU while doing pretty much nothing. > >> > >> Btw, compared with Asterisk, the latter is not even visible on the first > >> top's screen and spends 1% CPU when bridging two G711 calls and > recording > >> them to disk. > >> > >> So, at this time I have both original Asterisk and FS setups running. > One > >> is > >> seemless but clumsy in configuration, the other one is neat and stylish > >> but > >> too preoccupied with smth... Should I look into sofia epollers? That's > >> kind > >> of deep in the code. Or should I just stick with Asterisk? > >> > >> > >> > >> > >> > >> Anthony Minessale-2 wrote: > >> > > >> > There is another user here with a 300mhz box. I am willing to > >> investigate > >> > this improved performance for weak devices but I need to do it in a > >> sane > >> > cross-platform way. > >> > > >> > > >> > On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman > >> > >> >> wrote: > >> > > >> >> A word to the wise to the general FreeSWITCH community: If Anthony > >> >> Minessale suggests that you try to do any number of things, it's a > >> very > >> >> good idea to try all those ideas before continuing on. I've known > >> him, > >> >> MikeJ, and bkw for several years, and they almost always have very > >> good > >> >> ideas as to troubleshoot a problem in FreeSWITCH. It's extremely > >> >> frustrating to try to help people out who won't try the provided > >> >> suggestions first. > >> >> > >> >> And note directly to "eaf" - bogomips is quite possibly the least > >> >> significant bit of data about a cpu that you will get out of > >> >> /proc/cpuinfo... The name itself - bogo, means bogus. > >> >> http://en.wikipedia.org/wiki/Bogomips > >> >> > >> >> -Yossi > >> >> > >> >> _______________________________________________ > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > -- > >> > Anthony Minessale II > >> > > >> > FreeSWITCH http://www.freeswitch.org/ > >> > ClueCon http://www.cluecon.com/ > >> > Twitter: http://twitter.com/FreeSWITCH_wire > >> > > >> > AIM: anthm > >> > MSN:anthony_minessale at hotmail.com > >> > >< > >> MSN%3Aanthony_minessale at hotmail.com > > > > >> > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > >> > > > > >> > > >> > IRC: irc.freenode.net #freeswitch > >> > > >> > FreeSWITCH Developer Conference > >> > sip:888 at conference.freeswitch.org > >> > >< > >> sip%3A888 at conference.freeswitch.org > > > > >> > > >> > iax:guest at conference.freeswitch.org/888 > >> > > >> googletalk:conf+888 at conference.freeswitch.org > > > > >> > > > > >> > > >> > pstn:213-799-1400 > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> -- > >> View this message in context: > >> http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26678873.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://old.nabble.com/Choppy-sound-with-PCMU-tp26594250p26684048.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/625b8007/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:02:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:02:42 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call Message-ID: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/33d48ca5/attachment-0002.html From brian at freeswitch.org Mon Dec 7 14:12:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 16:12:59 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <6350BFD5-D6E4-42CB-AB46-7677F1B4D537@freeswitch.org> session:execute("start_dtmf"); /b On Dec 7, 2009, at 4:02 PM, Nik Middleton wrote: > Hi > > Is it possible to trap on DTMF on a bridged call within an LUA > script? I?ve tried setting the gateway to use inband, but no joy. > It looks like I could use start_dtmf, but I can?t see how to launch > this within LUA > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/af59690d/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 14:15:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 16:15:16 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> session:execute("start_dtmf"); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > > > Regards, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/adf1a629/attachment-0002.html From msc at freeswitch.org Mon Dec 7 14:18:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Dec 2009 14:18:20 -0800 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: Message-ID: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/ad082b09/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:56:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:56:16 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> References: <191c3a030912071415mdf25b80td9677f63b95bb433@mail.gmail.com> Message-ID: Once the call is bridged, while I can see an inband DTMF event being generated, it doesn't call my hook unfortuneately function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '*' then session:hangup(); return true; end session:execute("start_dtmf"); session:execute("bridge",bridgestring ); Am I missing something? Before the bridge, the oninput function works fine Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 22:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call session:execute("start_dtmf"); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0ce7a20d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 14:59:03 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 22:59:03 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: Can this be done in an lua script? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/23b7bc42/attachment-0002.html From Prometheus001 at gmx.net Mon Dec 7 15:12:29 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 08 Dec 2009 00:12:29 +0100 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <4B1AC410.9050201@gmx.net> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> <4B0C481A.8030309@gmx.net> <191c3a030911241359g1d48ec2foee56280c5a59a232@mail.gmail.com> <4B0C6499.4060504@gmx.net> <62CC2FF9-B45E-47AE-B0B8-2BA45B46B253@jerris.com> <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org> <4B0F1565.6060909@gmx.net> <191c3a030911271311q695b0829k580d1898610a4084@mail.gmail.com> <191c3a030911271322tce11dcy991dc1d668179a76@mail.gmail.com> <4B1AC410.9050201@gmx.net> Message-ID: <4B1D8BDD.7040505@gmx.net> Hello, i now changed the $${domain} of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Thanks to all for your support. Best regards Peter Peter P GMX schrieb: > Hello Anthony, > > I did some checks today > Here is how the phones are registered: > > mysql> select sip_host, presence_hosts, server_user,server_host, > hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip_host | presence_hosts | server_user | server_host | > hostname | sip_realm | mwi_user | mwi_host | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | > sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | > +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ > IPs are: > 10.11.12.1 sip1.mydomain.com (common cluster IP) > 10.11.12.2 sip11.mydomain.com > 10.11.12.3 sip12.mydomain.com (not used at this point) > > XML-Curl for the directory is: > >
> > > > > > > > > value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > >
> > > > The internal profile has the following alias: > > > > > > > With $${domain} being sip11.mydomain.com > > Phones are registering to sip1.mydomain.com, Voicemail works, but MWI > does not. Any hint what I should change to make this work? > > Best regards > Peter > > Anthony Minessale schrieb: > >> based on your example past >> >> sip1.mydomain.com is the domain in the >> packet and thus the profile should have an alias for this. >> Then the user must reside in your sip db with the user 200 and domain >> sip1.mydomain.com >> >> if you dont have this consider the force-register-domain and >> force-register-db-domain to normalize the host names. >> >> >> On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale >> > wrote: >> >> Did you check the 2 replies that told you you need aliases in your >> sofia profile to translate the domain found in your >> message_waiting to the right profile? Both Brian and Mike >> answered you. >> >> >> >> >> >> On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX >> > wrote: >> >> I tried now with phones directly attached to the freeswitch >> (without an >> OpenSIPS in between). I also added the alias. But the >> behaviour is as >> before: >> No notify message from freeswitch, neither after register nor >> after a >> voicemail is recorded. >> >> Best regards >> Peter >> Brian West schrieb: >> > Yes an alias will be required for every domain you run on >> the profile >> > so it can find it. >> > >> > /b >> > >> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> > >> > >> >> Try an alias on the sip profile. >> >> >> >> Mike >> >> >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > From larclap at yahoo.com Mon Dec 7 15:19:42 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 7 Dec 2009 15:19:42 -0800 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: <010001ca7793$c1e7c410$45b74c30$@com> It can. I use it like: session:execute("bind_meta_app", "1 b s execute_extension::dx XML features"); session:execute("bind_meta_app", "2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav"); session:execute("bind_meta_app", "3 b s execute_extension::cf XML features"); From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Monday, December 07, 2009 2:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call Can this be done in an lua script? Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0dd25521/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 15:21:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 17:21:22 -0600 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> Message-ID: <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Can this be done in an lua script? > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 07 December 2009 22:18 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Trapping dtmf on bridged call > > > > > > On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi > > > > Is it possible to trap on DTMF on a bridged call within an LUA script? > I?ve tried setting the gateway to use inband, but no joy. It looks like I > could use start_dtmf, but I can?t see how to launch this within LUA > > Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever > you want to have happen. Check it out: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app > > The Local_Extension in the default.xml dialplan file has a few examples of > using this tool. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/48514461/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Dec 7 15:32:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 7 Dec 2009 23:32:25 -0000 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> Message-ID: Yes I did, is it possible mod_vmd is interering? It's stopped before I call the start_dtmf function session:setHangupHook("myHangupHook", "blah") session:setInputCallback("onInput"); session:execute("vmd","start"); if (session:ready() == false) then freeswitch.consoleLog("info", " : Call Failed!!!\n"); end session:answer(); ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton wrote: Can this be done in an lua script? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/844a4375/attachment-0002.html From chris at fowler.cc Mon Dec 7 16:46:11 2009 From: chris at fowler.cc (Chris Fowler) Date: Mon, 7 Dec 2009 19:46:11 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <1260233114.13588.1348952845@webmail.messagingengine.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> Message-ID: <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> Skype have opened their beta program up to all comers. http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent Three lines in a sip_profile make FreeSWITCH talk nicely; but using the PCMU codec. Any progress on SILK native support? Last I saw was discussion back in September with Brian lamenting that Skype was hard to work with on this. I know I could use mod_skypiax; but having a native solution would be one less IT headache. Thx, Chris. From brian at freeswitch.org Mon Dec 7 17:27:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 19:27:46 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> Message-ID: <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> They have yet to type "make" on a 64bit box and build us a binary that is 64bit. Chances are they mucked it up like the BroadVoice codecs were and it just won't work on 64bit just yet... if they would just give us the src we could be done in under two days with it I suspect. /b On Dec 7, 2009, at 6:46 PM, Chris Fowler wrote: > Skype have opened their beta program up to all comers. > http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent > > Three lines in a sip_profile make FreeSWITCH talk nicely; but using > the > PCMU codec. > > Any progress on SILK native support? Last I saw was discussion back in > September with Brian lamenting that Skype was hard to work with on > this. > > I know I could use mod_skypiax; but having a native solution would be > one less IT headache. > > Thx, Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/4a531d2c/attachment-0002.html From jason at jasonjgw.net Mon Dec 7 17:39:38 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 8 Dec 2009 12:39:38 +1100 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> Message-ID: <20091208013938.GA4976@jdc.jasonjgw.net> Brian West wrote: > They have yet to type "make" on a 64bit box and build us a binary > that is 64bit. Chances are they mucked it up like the BroadVoice > codecs were and it just won't work on 64bit just yet... if they > would just give us the src we could be done in under two days with > it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) From brian at freeswitch.org Mon Dec 7 17:50:08 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 7 Dec 2009 19:50:08 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <20091208013938.GA4976@jdc.jasonjgw.net> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> Message-ID: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> it I suspect. > > Given that they released the codec specification, perhaps someone is > writing > an independent C implementation? (Not that I'm much interested, > but...) From mctch at yahoo.com Mon Dec 7 18:05:52 2009 From: mctch at yahoo.com (Mark Crane) Date: Mon, 7 Dec 2009 18:05:52 -0800 (PST) Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1D78AF020000E1000002BC@mail.fribert.dk> Message-ID: <659603.29094.qm@web56408.mail.re3.yahoo.com> Question ---------------------------------------------- I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- Since you moved the internal profile to the lan ip address you can go ahead and dump the lan profile. Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer --- On Mon, 12/7/09, mailinglist wrote: From: mailinglist Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Monday, December 7, 2009, 1:50 PM Hi All ? Ok, so next episode in the saga of getting this monster of the ground :-) ? I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. ? If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] ? I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? ? ? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name??????????????????? internal Domain Name???????????? N/A DBName????????????????? sofia_reg_internal Pres Hosts Dialplan??????????????? XML Context???????????????? public Challenge Realm???????? auto_from RTP-IP????????????????? 10.11.12.25 Ext-RTP-IP????????????? 10.11.12.25 SIP-IP????????????????? 10.11.12.25 Ext-SIP-IP????????????? 10.11.12.25 URL???????????????????? sip:mod_sofia at 10.11.12.25:5060 BIND-URL??????????????? sip:mod_sofia at 10.11.12.25:5060 HOLD-MUSIC????????????? local_stream://moh OUTBOUND-PROXY????????? N/A CODECS????????????????? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT?????????????? 101 DTMF-MODE?????????????? rfc2833 CNG???????????????????? 13 SESSION-TO????????????? 0 MAX-DIALOG????????????? 0 NOMEDIA???????????????? false LATE-NEG??????????????? false PROXY-MEDIA???????????? false AGGRESSIVENAT?????????? false STUN-ENABLED??????????? true STUN-AUTO-DISABLE?????? false CALLS-IN??????????????? 10 FAILED-CALLS-IN???????? 5 CALLS-OUT?????????????? 0 FAILED-CALLS-OUT??????? 0 ? Registrations: ================================================================================================= Call-ID:??????? MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User:?????????? 1001 at 10.11.12.25 Contact:??????? "1001" Agent:????????? X-Lite release 1103k stamp 53621 Status:???????? Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host:?????????? firewall.fribert.dk IP:???????????? 10.11.12.145 Port:?????????? 59650 Auth-User:????? 1001 Auth-Realm:???? 10.11.12.25 ? Call-ID:??????? OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User:?????????? 1002 at 10.11.12.25 Contact:??????? "1002" Agent:????????? 3CXVoipPhone 3.1.6288.0 Status:???????? Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host:?????????? firewall.fribert.dk IP:???????????? 10.11.12.195 Port:?????????? 4117 Auth-User:????? 1002 Auth-Realm:???? 10.11.12.25 ? ================================================================================================= ? And... ? sofia status profile external API CALL [sofia(status profile external)] output: ================================================================================================= Name??????????????????? external Domain Name???????????? N/A DBName????????????????? sofia_reg_external Pres Hosts Dialplan??????????????? XML Context???????????????? public Challenge Realm???????? auto_to RTP-IP????????????????? 87.61.18.196 Ext-RTP-IP????????????? 87.61.18.196 SIP-IP????????????????? 87.61.18.196 Ext-SIP-IP????????????? 87.61.18.196 URL???????????????????? sip:mod_sofia at 87.61.18.196:5080 BIND-URL??????????????? sip:mod_sofia at 87.61.18.196:5080 HOLD-MUSIC????????????? local_stream://moh OUTBOUND-PROXY????????? N/A CODECS????????????????? PCMU,PCMA,GSM TEL-EVENT?????????????? 101 DTMF-MODE?????????????? rfc2833 CNG???????????????????? 13 SESSION-TO????????????? 0 MAX-DIALOG????????????? 0 NOMEDIA???????????????? false LATE-NEG??????????????? false PROXY-MEDIA???????????? false AGGRESSIVENAT?????????? false STUN-ENABLED??????????? true STUN-AUTO-DISABLE?????? false CALLS-IN??????????????? 0 FAILED-CALLS-IN???????? 0 CALLS-OUT?????????????? 2 FAILED-CALLS-OUT??????? 2 ? Registrations: ================================================================================================= ================================================================================================= ? ? In my Dialplan I've created these two entries: ? ----- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 ----- ? and ? ----- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$1 ? I'm not sure if the latter is correct or needed to make local calls? But anyways, it doesn't seem to react as per my intentions. ? If I try and make a local call from 1001 to 1002 it says ?2009-12-07 21:40:02.776210 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b115e2b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.776210 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-07 21:40:02.796449 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [aa22e3b1-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:40:02.874492 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATION 2009-12-07 21:40:02.894599 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/external/$1) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1085 Session 14 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-07 21:40:02.894599 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] ? As I read it it goes into context default, and then tries to create an external channel, which I don't understand why? And then it fails of course. ? Then if I try to do an external call (with the leading 0) it gives me: 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.1.25 [25260ce8-70e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.655915 [INFO] mod_dialplan_xml.c:252 Processing 1001->012345678 in context dfault 2009-12-07 21:41:33.655915 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [36e10ce870e3-de11-af59-000c29b7b4cb] 2009-12-07 21:41:33.755921 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NOROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATON 2009-12-07 21:41:33.755921 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [C_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 17 (sofia/external/$1) Ened 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [S_DESTROY] 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/internal/1001 at 1.11.12.25) Ended 2009-12-07 21:41:33.755921 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/100110.11.12.25 [CS_DESTROY] So for me, it looks like it never comes to the dialplan I've entered into the pfsense interface??? I've used the gateway value instead of the profile value in my bridge. So the question is, do I go and enter the 'default.xml' for the dialplan, or what do I do? What have I missed here??? ? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/0411e0d7/attachment-0002.html From djbinter at yahoo.com Mon Dec 7 18:29:12 2009 From: djbinter at yahoo.com (DJB) Date: Mon, 7 Dec 2009 18:29:12 -0800 (PST) Subject: [Freeswitch-users] Zombie Records in core db Message-ID: <151010.8763.qm@web37501.mail.mud.yahoo.com> We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from "show calls" that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those "zombied" records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/306f7071/attachment-0002.html From pete at privateconnect.com Mon Dec 7 19:16:47 2009 From: pete at privateconnect.com (Pete Mueller) Date: Mon, 7 Dec 2009 20:16:47 -0700 Subject: [Freeswitch-users] Trapping dtmf on bridged call In-Reply-To: References: <87f2f3b90912071418l6d2f774do51f41a93a8297b2@mail.gmail.com> <191c3a030912071521i73d5ae07tb6da5d8a9d5c820d@mail.gmail.com> Message-ID: <002401ca77b4$e0d64d80$a282e880$@com> I had this featured requested of me a few months ago. Bind_meta_app does work, but requires two tones, the "*" plus an additional digit. I needed to perform a task on the "*". I re-wrote the bind_meta_app handler so that if you attached a instruction to what would be "**" hitting a single "*" would active it. Kind of a hack, but if no one comes up with a more elegant way, I could provide a .patch file that did the necessary changes. I'd love an all-LUA method, or something that could use an existing InputCallback routine, but this worked for my immediate need. -pete From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Monday, December 07, 2009 4:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call Yes I did, is it possible mod_vmd is interering? It's stopped before I call the start_dtmf function session:setHangupHook("myHangupHook", "blah") session:setInputCallback("onInput"); session:execute("vmd","start"); if (session:ready() == false) then freeswitch.consoleLog("info", " : Call Failed!!!\n"); end session:answer(); _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton wrote: Can this be done in an lua script? Regards, _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/5a732875/attachment-0002.html From dujinfang at gmail.com Mon Dec 7 19:33:27 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 8 Dec 2009 11:33:27 +0800 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <23f91030912071933w231c61e6gcd4e2aa555f2794d@mail.gmail.com> I also have this problem on a trunk version more than 1000 revisions behind, so I think the best way is to upgrade to trunk and report this again if still have problem. 2009/12/8 DJB : > We have FreeSWITCH Version 1.0.4 (exported) running at a high volume > traffic. ?I normally check the concurrent calls by looking at the number of > sessions from status command. ?However, the number of concurrent calls in FS > is normally higher than it's supposed to be after we ran traffic for about a > week. ?Thus, I routed the traffic away from the FS and found out from "show > calls" that there were so many old calls from previous days. ?We are running > a pass-thru traffic in signaling only. ?I wonder whether there is a way to > have those "zombied" records clean up automatically. ?Also, what should I do > to prevent this problem? > Thank you, > Dorn B. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Mon Dec 7 19:41:03 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 7 Dec 2009 19:41:03 -0800 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <580F5165-09AB-44DF-B857-E9B1A43D076D@freeswitch.org> Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk or at least pre8 when it's available. -MC Sent from my iPhone On Dec 7, 2009, at 6:29 PM, DJB wrote: > We have FreeSWITCH Version 1.0.4 (exported) running at a high volume > traffic. I normally check the concurrent calls by looking at the > number of sessions from status command. However, the number of > concurrent calls in FS is normally higher than it's supposed to be > after we ran traffic for about a week. Thus, I routed the traffic > away from the FS and found out from "show calls" that there were so > many old calls from previous days. We are running a pass-thru > traffic in signaling only. I wonder whether there is a way to have > those "zombied" records clean up automatically. Also, what should I > do to prevent this problem? > > Thank you, > Dorn B. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/dc872afe/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 7 19:47:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Dec 2009 21:47:01 -0600 Subject: [Freeswitch-users] Zombie Records in core db In-Reply-To: <151010.8763.qm@web37501.mail.mud.yahoo.com> References: <151010.8763.qm@web37501.mail.mud.yahoo.com> Message-ID: <191c3a030912071947q3dd98adbn320b2a1b7f1b25bc@mail.gmail.com> For starters, try using the latest svn snapshot. Your version is 6 months old and several thousand revs old. On Dec 7, 2009 8:34 PM, "DJB" wrote: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from "show calls" that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those "zombied" records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091207/91a99bbe/attachment-0002.html From mailinglist at fribert.dk Mon Dec 7 22:05:31 2009 From: mailinglist at fribert.dk (mailinglist) Date: Tue, 08 Dec 2009 07:05:31 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <659603.29094.qm@web56408.mail.re3.yahoo.com> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> Message-ID: <4B1DFABC020000E1000002C2@mail.fribert.dk> Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/20d25408/attachment-0002.html From yehavi.bourvine at gmail.com Mon Dec 7 22:50:11 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Dec 2009 08:50:11 +0200 Subject: [Freeswitch-users] Translating DTMF from RFC2833 to INFO In-Reply-To: <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> References: <4468a6770912030043j5212663cxab879d738da391ad@mail.gmail.com> <4B1B865A.2060901@metik.com> <4B1C031B.8060906@metik.com> <191c3a030912061135g13ad2f48kfe8f935804b1fae@mail.gmail.com> Message-ID: Hello all, *debug voip rtp session named-event*s shows that it receives and understands the DTMFs, but it does not send them to the PSTN (sends only those received via INFO). I haveto find some time and go to the remote site to update to the latest IOS... I will update after this has been done. Regards, __Yehavi: 2009/12/6 Anthony Minessale > Some more bad news for you, info dtmf spec has expired and has been > abandoned. Wait till you see what they did accept instead...... > > On Dec 6, 2009 1:22 PM, "Metik" wrote: > > Unless the IOS you are running is extremely buggy, "debug voip ccapi" > commands should not provide you with that detail, what you really want > to use is "debug voip rtp session named-event". > > Normal SIP-to-PSTN calls should use both a pots and voip dial peer but > DTMF relay type is determined by the voip dial peer. > > I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) > previously in the wild. Unlike some other SIP feature servers, I have > not had issues (with RFC 2833) between FS and Cisco IOS gateways. > > Although unrelated to FS or any other SIP feature server, I have seen > some issues when multple dtmf relay types are left enabled on a voip > dial peer. Also, there are some (older) IOS versions that have issues > with DTMF duration which cause digits to be misinterpreted by the > far-end (PSTN/POTS) but not ignored altogether. > > -metik > > Yehavi Bourvine wrote: > Hello Metik, > > > > 2009/12/6 Metik < > freeswitch-users-list at metik.com > > > > > > > You previously stated that your Cisco gateway has some "bug" that > > prevents you from us... > > > ------------------------------------------------------------------------ > > > _____________________... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e07a31a9/attachment-0002.html From jingwei.yang at gmail.com Tue Dec 8 01:09:29 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 8 Dec 2009 17:09:29 +0800 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> Message-ID: <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> Hi Jo?o, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. Here's how I normally update FreeSwitch: *make clean && svn up && ./bootstrap.sh && ./configure && make install * If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. Regards, -Jingwei 2009/12/8 Jo?o Mesquita > Maybe, just maybe isse that make target to reconf libtiff? > > Regards, > > JM > > > On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > >> I installed libjpeg-7 following this website: >> http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And >> the previous error is replaced by a new one: >> >> gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. >> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF >> .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o >> at_interpreter.o >> at_interpreter.c: In function ???command_search???: >> at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use >> in this function) >> at_interpreter.c:5299: error: (Each undeclared identifier is reported only >> once >> at_interpreter.c:5299: error: for each function it appears in.) >> at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in >> this function) >> at_interpreter.c: In function ???at_interpreter???: >> at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in >> this function) >> make[8]: *** [at_interpreter.lo] Error 1 >> >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> However, I'm still able to start freeswitch and mod_skypiax and make skype >> calls with no problem. >> >> Regards, >> -Jingwei >> >> >> >> On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: >> >>> No, I didn't change or update the system libs. I just wanted to double >>> check whether my system has this libjpeg library. ./configure was definitely >>> executed before the source codes were rebuilt. >>> >>> Regards, >>> -Jingwei >>> >>> >>> On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: >>> >>>> Hi, >>>> >>>> That one is on your side. If you changed/updated system libs it might be >>>> worth doing another ./configure >>>> >>>> Cheers, >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: >>>> >>>> Hi Mathieu, thanks for the promptly reply. The error has been fixed. >>>> However, I encounter another one. >>>> >>>> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 >>>> -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>>> -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >>>> -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o >>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>>> -lc >>>> ./make_at_dictionary: error while loading shared libraries: >>>> libjpeg.so.7: cannot open shared object file: No such file or directory >>>> make[8]: *** [at_interpreter_dictionary.h] Error 127 >>>> make[7]: *** [all] Error 2 >>>> make[6]: *** [all-recursive] Error 1 >>>> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >>>> make[4]: *** [install] Error 1 >>>> make[3]: *** [mod_voipcodecs-install] Error 1 >>>> make[2]: *** [install-recursive] Error 1 >>>> >>>> Do you have idea about this one? >>>> >>>> Thanks! >>>> >>>> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >>>> >>>>> Consider it fixed. >>>>> Committed revision 15765. >>>>> >>>>> Mathieu Rene >>>>> Avant-Garde Solutions Inc >>>>> Office: + 1 (514) 664-1044 x100 >>>>> Cell: +1 (514) 664-1044 x200 >>>>> mrene at avgs.ca >>>>> >>>>> >>>>> >>>>> >>>>> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >>>>> >>>>> Hi Guys, >>>>> >>>>> I got a compilation error of skypiax_protocol.c with the latest version >>>>> r15764. >>>>> >>>>> Compiling skypiax_protocol.c... >>>>> *cc1: warnings being treated as errors* >>>>> skypiax_protocol.c: In function ???X11_errors_handler???: >>>>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c: In function ???skypiax_send_message???: >>>>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>>>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations >>>>> and code >>>>> make[5]: *** [skypiax_protocol.o] Error 1 >>>>> make[4]: *** [install] Error 1 >>>>> make[3]: *** [mod_skypiax-install] Error 1 >>>>> make[2]: *** [install-recursive] Error 1 >>>>> >>>>> I personally checked the file and it shouldn't be a merge problem. Does >>>>> anyone encounter this as well? >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/cb574ba9/attachment-0002.html From jingwei.yang at gmail.com Tue Dec 8 02:01:57 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 8 Dec 2009 18:01:57 +0800 Subject: [Freeswitch-users] OpenZap issues with incoming and outgoing calls In-Reply-To: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> References: <13529f9d0912030129l3f7be0adke1af5fd7f55cb069@mail.gmail.com> Message-ID: <13529f9d0912080201s8f6db58w7fafc3a41de3739f@mail.gmail.com> Problem solved. It's due to the lack of definition in tones.conf. In case anyone else need it, here's the tone plan for Singapore. [sg] generate-dial => v=-7;%(1000,0,425) detect-dial => 425 generate-ring => v=-7;%(2000,4000,425) detect-ring => 425 generate-busy => v=-7;%(750,750,425) detect-busy => 425 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 1776.7 On Thu, Dec 3, 2009 at 5:29 PM, Jingwei Yang wrote: > Hello All, > > I have a Digium TDM400P pci card with two FXO ports installed on my linux > box. I've connected an external telephone line to the first FXO port. But I > can't either make outgoing calls or receive incoming ones. Here are my > setups, please let me know where goes wrong. > * > /etc/zaptel.conf* > > loadzone = sg > defaultzone=sg > fxsks=1,2 > > */usr/local/freeswitch/conf/zt.conf* remains unchanged > > [defaults] > codec_ms => 20 > wink_ms => 150 > flash_ms => 750 > echo_cancel_level => 64 > rxgain => 0.0 > txgain => 0.0 > > */usr/local/freeswitch/conf/openzap.conf* > > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > [span zt] > name => OpenZAP > number => 2 > fxo-channel => 2 > > */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* > > > > > > > > > > > > > > > > > > > > > > > > > I defined an extension in dialplan/default.xml to receive bridge incoming > calls to my skype instance. Frankly speaking, I'm not sure whether this > definition is correct. How should I define the expression? When I dial the > telephone number, the FS console has no response and I hear nother but busy > tones. > > > > > > > > For outgoing calls, I tried something like this: originate > openzap/1/1/xxxxxxxx &echo, while "xxxxxxxx" is my handphone number. Again, > my handphone has no response. Hopefully I've explained my situation clearly. > Please kindly enlighten where the problem might be. > > Thanks, > -Jingwei > > p.s. here is the outgoing log trace for your reference. > > > freeswitch at localhost.localdomain> originate openzap/1/1/xxxxxxxx &echo > 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 > taps for 1:1 > 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound > channel OpenZAP/1:1/xxxxxxxx > 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel > OpenZAP/1:1/xxxxxxxx [6f843194-18ce-4525-862f-f5f4e96db5eb] > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 > (OpenZAP/1:1/xxxxxxxx) State Change CS_NEW -> CS_INIT > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 > from DOWN to DIALING > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread > starting. > 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 > taps for 1:1 > 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for DIALING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_INIT > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/1:1/xxxxxxxx) State INIT > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/xxxxxxxx) > State Change CS_INIT -> CS_ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 > (OpenZAP/1:1/xxxxxxxx) State INIT going to sleep > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/1:1/xxxxxxxx) State ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/xxxxxxxx > CHANNEL ROUTING > 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 > (OpenZAP/1:1/xxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 > (OpenZAP/1:1/xxxxxxxx) State ROUTING going to sleep > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA > 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:360 > (OpenZAP/1:1/xxxxxxxx) State CONSUME_MEDIA going to sleep > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_HOME ||| > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_OFFICE ||| > 2009-12-03 17:21:34.114940 [DEBUG] skypiax_protocol.c:141 rev > 15765M[(nil)|37 ][DEBUG_SKYPE 141 ][skypiax8 ][-1, 0, 0] READING: > |||USER amanda8884 PHONE_MOBILE ||| > 2009-12-03 17:21:34.684709 [DEBUG] ozmod_analog.c:340 Changing state on 1:1 > from DIALING to BUSY > 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for BUSY > 2009-12-03 17:21:34.704705 [DEBUG] ozmod_analog.c:579 Changing state on 1:1 > from BUSY to DOWN > 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:450 Executing state > handler on 1:1 for DOWN > 2009-12-03 17:21:34.724706 [DEBUG] mod_openzap.c:1334 got FXO sig 1:1 > [STOP] > 2009-12-03 17:21:34.724706 [NOTICE] mod_openzap.c:1352 Hangup > OpenZAP/1:1/xxxxxxxx [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] > 2009-12-03 17:21:34.724706 [DEBUG] switch_channel.c:1912 Send signal > OpenZAP/1:1/xxxxxxxx [KILL] > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_HANGUP > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:459 thread > mismatch skipping state handler. > 2009-12-03 17:21:34.724706 [DEBUG] zap_io.c:1234 channel done 1:1 > 2009-12-03 17:21:34.724706 [DEBUG] ozmod_analog.c:766 ANALOG CHANNEL 1:1 > thread ended. > 2009-12-03 17:21:34.724706 [DEBUG] switch_core_state_machine.c:486 > (OpenZAP/1:1/xxxxxxxx) State HANGUP > API CALL [originate(openzap/1/1/xxxxxxxx &echo)] output: > -ERR NORMAL_CIRCUIT_CONGESTION > > 2009-12-03 17:21:34.724706 [DEBUG] switch_ivr_originate.c:2988 Originate > Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] > 2009-12-03 17:21:34.725762 [WARNING] mod_openzap.c:474 VETO Changing state > on 1:1 from DOWN to HANGUP > 2009-12-03 17:21:34.725762 [DEBUG] mod_openzap.c:510 OpenZAP/1:1/xxxxxxxx > CHANNEL HANGUP > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:46 > OpenZAP/1:1/xxxxxxxx Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:486 > (OpenZAP/1:1/xxxxxxxx) State HANGUP going to sleep > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:333 > (OpenZAP/1:1/xxxxxxxx) State Change CS_HANGUP -> CS_REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:314 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 > (OpenZAP/1:1/xxxxxxxx) State REPORTING > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:53 > OpenZAP/1:1/xxxxxxxx Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:577 > (OpenZAP/1:1/xxxxxxxx) State REPORTING going to sleep > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:327 > (OpenZAP/1:1/xxxxxxxx) State Change CS_REPORTING -> CS_DESTROY > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:999 Send signal > OpenZAP/1:1/xxxxxxxx [BREAK] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_session.c:1136 Session 1 > (OpenZAP/1:1/xxxxxxxx) Locked, Waiting on external entities > 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1154 Session 1 > (OpenZAP/1:1/xxxxxxxx) Ended > 2009-12-03 17:21:34.725762 [NOTICE] switch_core_session.c:1156 Close > Channel OpenZAP/1:1/xxxxxxxx [CS_DESTROY] > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:423 > (OpenZAP/1:1/xxxxxxxx) Running State Change CS_DESTROY > 2009-12-03 17:21:34.725762 [DEBUG] switch_core_state_machine.c:434 > (OpenZAP/1:1/xxxxxxxx) State DESTROY > freeswitch at localhost.localdomain> 2009-12-03 17:21:34.726741 [DEBUG] > switch_core_state_machine.c:60 OpenZAP/1:1/xxxxxxxx Standard DESTROY > 2009-12-03 17:21:34.726741 [DEBUG] switch_core_state_machine.c:434 > (OpenZAP/1:1/xxxxxxxx) State DESTROY going to sleep > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c98e19a7/attachment-0002.html From gmaruzz at celliax.org Tue Dec 8 02:14:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 8 Dec 2009 11:14:36 +0100 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> Message-ID: <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > We can ONLY hope someone will do this and BSD/MIT the library and NOT > GPL it... if they GPL it then we'll have to have someone write it all > over again... love the Open Source oil and water. > > /b > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > >>> it I suspect. >> >> Given that they released the codec specification, perhaps someone is >> writing >> an independent C implementation? (Not that I'm much interested, >> but...) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From yehavi.bourvine at gmail.com Tue Dec 8 03:12:37 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Dec 2009 13:12:37 +0200 Subject: [Freeswitch-users] Debugging reeswitch (especially TLS) Message-ID: Hello, I have some black hole understading how to debug Freeswitch. In fs_cli I do "sofia debug all 7" and indeed get a lot of debugging messages on the console; however, the logfiles get only Critical messages. Where do I define which messages go to the logfile? And in a related topic: I've set a Polycom to use TLS with Freeswitch. I see it contacts FS on TCP port 5061, do some exchange, and then quits and does not use TLS. How do I debug TLS from FS side? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/106bb37f/attachment-0002.html From jbr at consiglia.dk Tue Dec 8 03:46:26 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 8 Dec 2009 12:46:26 +0100 Subject: [Freeswitch-users] Lua and database access to core_db Message-ID: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not "like": require "luasql.sqlite" env = luasql.sqlite() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) After changing it to require "luasql.sqlite3" env = luasql.sqlite3() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/4764dc8a/attachment-0002.html From codecomplete at free.fr Tue Dec 8 05:43:25 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 05:43:25 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? Message-ID: <26694069.post@talk.nabble.com> Hello I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm getting a software error althought the TDM card seems detected (lspci -v OK). Dahdi was successfully compiled from source code. Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead with OpenZAP? The reason I ask, is that in another forum, someone mentionned "/etc/asterisk/chan_dahdi.conf". Here's the output from dahdi_cfg -vvv: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Here's ls -l /dev/dahdi/: total 0 crw-rw---- 1 root root 196, 254 Dec 8 13:38 channel crw-rw---- 1 root root 196, 0 Dec 8 13:38 ctl crw-rw---- 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw---- 1 root root 196, 253 Dec 8 13:38 timer Has someone succesfully installed Dahdi without Asterisk? Thank you. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694069.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Russell.Mosemann at cune.org Tue Dec 8 06:15:36 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:15:36 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694069.post@talk.nabble.com> Message-ID: <20091208141536.8AD6E421BDE@mail.cune.org> Fred-145 said: > Has someone succesfully installed Dahdi without Asterisk? Of course, and it's working like a charm. DAHDI is a driver. It doesn't care what software uses it. We're using DAHDI with a TE110P PRI T1 card. What is in /proc/dahdi? If it shows "1", what do you see if you "cat /proc/dahdi/1"? Did you correctly configure the files in /etc/dahdi? How did you configure ../freeswitch/conf/openzap.conf? Maybe it would be helpful to spend a few minutes browsing the wiki at http://wiki.freeswitch.org/wiki/OpenZAP -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From kevin at johnnyvoip.com Tue Dec 8 06:17:15 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Tue, 8 Dec 2009 09:17:15 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Message-ID: It seems you can get a copy of either the binaries or the source by doing the following: - Review & execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. - Return executed agreement to silksupport at skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN - Skype will email you the SILK binary once we receive the executed agreement. - Check out documentation, FAQ, and discussion forum (URL TBD) - Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli wrote: > Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... > > On Tue, Dec 8, 2009 at 2:50 AM, Brian West wrote: > > We can ONLY hope someone will do this and BSD/MIT the library and NOT > > GPL it... if they GPL it then we'll have to have someone write it all > > over again... love the Open Source oil and water. > > > > /b > > > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > > > >>> it I suspect. > >> > >> Given that they released the codec specification, perhaps someone is > >> writing > >> an independent C implementation? (Not that I'm much interested, > >> but...) > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/95d9c25b/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Copy of Skype - SILK Codec License 27052009.pdf Type: application/pdf Size: 51837 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/95d9c25b/attachment-0002.pdf From mike at jerris.com Tue Dec 8 06:27:24 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 09:27:24 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> Message-ID: <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > It seems you can get a copy of either the binaries or the source by > doing the following: > > Review & execute SILK Agreement - attached. NOTE - please add your > Skype login to this form also. > Return executed agreement to silksupport at skype.net and mail hardcopy > to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN > Skype will email you the SILK binary once we receive the executed > agreement. > Check out documentation, FAQ, and discussion forum (URL TBD) > Provide feedback to Skype. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > wrote: > Or it can be LGPL, that's acceptable for FreeSWITCH for my > understanding... > > On Tue, Dec 8, 2009 at 2:50 AM, Brian West > wrote: > > We can ONLY hope someone will do this and BSD/MIT the library and > NOT > > GPL it... if they GPL it then we'll have to have someone write it > all > > over again... love the Open Source oil and water. > > > > /b > > > > On Dec 7, 2009, at 7:39 PM, Jason White wrote: > > > >>> it I suspect. > >> > >> Given that they released the codec specification, perhaps someone > is > >> writing > >> an independent C implementation? (Not that I'm much interested, > >> but...) > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/66387092/attachment-0002.html From codecomplete at free.fr Tue Dec 8 06:33:17 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 06:33:17 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <20091208141536.8AD6E421BDE@mail.cune.org> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> Message-ID: <26694801.post@talk.nabble.com> Thanks Russel for the tip. After more googling, I ended up figuring that /etc/dahdi/modules had to contain the list of drivers to load. For those interested, here's how to compile and install Dahdi (which doesn't need Asterisk at all, unlike some docs on the Net seem to imply due to references to /etc/asterisk/*.conf): 1. Download and unpack the Dahdi tarball 2. make all ; make install ; make config 3. cd /etc/dahdi/ 4. vi system.conf: #For France, single FXO module on TDM card loadzone = fr defaultzone = fr fxsks=1 5. vi modules: wcfxo wctdm dahdi 6. /etc/init.d/dahdi start 7. dahdi_cfg -vvv 8. ls -la /proc/dahdi/ Now, on to OpenZAP... Thanks again. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694801.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 8 06:33:55 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 08:33:55 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: > Changing the core db into a MySQL via ODBC caused some problems even > after it seemed to work. For instance, console help caused an error > with an error description indicating that a SQL SELECT query > including the reserved word key has been fired. > > It this problem likely to be solved if I used another version of the > MySQL? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/042c007c/attachment-0002.html From kevin at johnnyvoip.com Tue Dec 8 06:39:22 2009 From: kevin at johnnyvoip.com (Kevin Green) Date: Tue, 8 Dec 2009 09:39:22 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > > It seems you can get a copy of either the binaries or the source by doing > the following: > > > - Review & execute SILK Agreement - attached. NOTE - please add your > Skype login to this form also. > - Return executed agreement to > silksupport at skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd > Floor, 2 Stephen Street, London W1T 1AN > - Skype will email you the SILK binary once we receive the executed > agreement. > - Check out documentation, FAQ, and discussion forum (URL TBD) > - Provide feedback to Skype. > > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli < > gmaruzz at celliax.org> wrote: > >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West < >> brian at freeswitch.org> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and NOT >> > GPL it... if they GPL it then we'll have to have someone write it all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > >> FreeSWITCH-users at lists.freeswitch.org >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9566f1ad/attachment-0002.html From Russell.Mosemann at cune.org Tue Dec 8 06:44:40 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:44:40 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> Message-ID: <20091208144440.4092839B743@mail.cune.org> Fred-145 said: > 5. vi modules: > wcfxo > wctdm > dahdi You only need one of the modules above, if you have one card. I don't see a "dahdi" module listed in the file here. > 8. ls -la /proc/dahdi/ You should be able to cat the file in that directory for more information. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From Russell.Mosemann at cune.org Tue Dec 8 06:46:06 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 14:46:06 -0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> Message-ID: <20091208144606.18CF03E57BD@mail.cune.org> Fred-145 said: > For those interested, here's how to compile and install Dahdi It would be helpful to others if you add the results of your efforts to the wiki. -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From codecomplete at free.fr Tue Dec 8 07:32:42 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 8 Dec 2009 07:32:42 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <20091208144440.4092839B743@mail.cune.org> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> <20091208144440.4092839B743@mail.cune.org> Message-ID: <26695674.post@talk.nabble.com> Russell.Mosemann wrote: > You only need one of the modules above, if you have one card. I don't see > a "dahdi" module listed in the file here. Yup, turns out wcfxo is needed for the X10xP card, while wctdm is needed for Digium cards. As for dahdi, maybe wcfxo/wctdm loads the dahdi module automatically? Russell.Mosemann wrote: > 8. ls -la /proc/dahdi/ You should be able to cat the file in that > directory for more information. Yes indeed: # ls -al /proc/dahdi/ total 0 dr-xr-xr-x 2 root root 0 Dec 8 16:30 . dr-xr-xr-x 80 root root 0 Dec 8 13:37 .. -r--r--r-- 1 root root 0 Dec 8 16:30 1 # cat 1 Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 3 WCTDM/4/2 4 WCTDM/4/3 Thanks for the tip. I'll see if I can update the wiki accordingly. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26695674.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mrene_lists at avgs.ca Tue Dec 8 07:49:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Dec 2009 10:49:32 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: They provide you with a 32 bit library, with the header files to link with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 8-Dec-09, at 9:39 AM, Kevin Green wrote: > Their site (https://developer.skype.com/silk) specifies that they > will provide the source, which as you say may not be 64-Bit > compatible but could likely be tweaked to work. I think you just > need to be specific in that you want a source copy not a binary copy > of the codec. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris > wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > >> It seems you can get a copy of either the binaries or the source by >> doing the following: >> >> Review & execute SILK Agreement - attached. NOTE - please add your >> Skype login to this form also. >> Return executed agreement to silksupport at skype.net and mail >> hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, >> London W1T 1AN >> Skype will email you the SILK binary once we receive the executed >> agreement. >> Check out documentation, FAQ, and discussion forum (URL TBD) >> Provide feedback to Skype. >> >> Regards, >> Kevin Green >> >> JohnnyVoIP >> http://www.johnnyvoip.com >> >> >> On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > > wrote: >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West >> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and >> NOT >> > GPL it... if they GPL it then we'll have to have someone write it >> all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps >> someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ad3fe9fd/attachment-0002.html From mike at jerris.com Tue Dec 8 07:58:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 10:58:53 -0500 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> We have as of yet been unable to obtain source and we have been in very close contact with skype all the way up to the lead technical and business people on this project. We would of course welcome access to the source but we have as of yet not been able to get a copy Mike On Dec 8, 2009, at 9:39 AM, Kevin Green wrote: > Their site (https://developer.skype.com/silk) specifies that they > will provide the source, which as you say may not be 64-Bit > compatible but could likely be tweaked to work. I think you just > need to be specific in that you want a source copy not a binary copy > of the codec. > > Regards, > Kevin Green > > JohnnyVoIP > http://www.johnnyvoip.com > > > On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris > wrote: > That would binary only, not 64 bit Linux . > > On Dec 8, 2009, at 9:17 AM, Kevin Green wrote: > >> It seems you can get a copy of either the binaries or the source by >> doing the following: >> >> Review & execute SILK Agreement - attached. NOTE - please add your >> Skype login to this form also. >> Return executed agreement to silksupport at skype.net and mail >> hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, >> London W1T 1AN >> Skype will email you the SILK binary once we receive the executed >> agreement. >> Check out documentation, FAQ, and discussion forum (URL TBD) >> Provide feedback to Skype. >> >> Regards, >> Kevin Green >> >> JohnnyVoIP >> http://www.johnnyvoip.com >> >> >> On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli > > wrote: >> Or it can be LGPL, that's acceptable for FreeSWITCH for my >> understanding... >> >> On Tue, Dec 8, 2009 at 2:50 AM, Brian West >> wrote: >> > We can ONLY hope someone will do this and BSD/MIT the library and >> NOT >> > GPL it... if they GPL it then we'll have to have someone write it >> all >> > over again... love the Open Source oil and water. >> > >> > /b >> > >> > On Dec 7, 2009, at 7:39 PM, Jason White wrote: >> > >> >>> it I suspect. >> >> >> >> Given that they released the codec specification, perhaps >> someone is >> >> writing >> >> an independent C implementation? (Not that I'm much interested, >> >> but...) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >> freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e0fbd696/attachment-0002.html From Prometheus001 at gmx.net Tue Dec 8 08:02:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 08 Dec 2009 17:02:28 +0100 Subject: [Freeswitch-users] Force presence status manually Message-ID: <4B1E7894.2050101@gmx.net> Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. Best regards Peter From spencer at 5ninesolutions.com Mon Dec 7 13:34:05 2009 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 7 Dec 2009 13:34:05 -0800 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers Message-ID: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. From brian at freeswitch.org Tue Dec 8 08:04:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 10:04:32 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> Message-ID: The fun part comes when you try to link that 32bit .a file into a 64bit so file. :P /b On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote: > They provide you with a 32 bit library, with the header files to > link with it. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/220d3e6d/attachment-0002.html From brian at freeswitch.org Tue Dec 8 08:04:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 10:04:44 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> References: <1260233114.13588.1348952845@webmail.messagingengine.com> <7454A296C7EDE34EA57199FAA401E2F11B7E074A06@VMBX113.ihostexchange.net> <09EB9105-E094-4947-9C64-51AABDE4E619@freeswitch.org> <20091208013938.GA4976@jdc.jasonjgw.net> <7b197bef0912080214k4a8645aby150d9d4454d6c2e@mail.gmail.com> <8FD974CE-F6D1-4B51-8051-B1D03873C955@jerris.com> <51A0820B-2073-4FF4-A6ED-D4E11B005226@jerris.com> Message-ID: I have resubmitted our request for the source. /b On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote: > We have as of yet been unable to obtain source and we have been in > very close contact with skype all the way up to the lead technical > and business people on this project. We would of course welcome > access to the source but we have as of yet not been able to get a copy > > Mike From mike at jerris.com Tue Dec 8 08:09:53 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:09:53 -0500 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: <5FA1ADE4-ECF9-4C33-A2FE-2EACB8A946AE@jerris.com> You definitely need to use the settings in combination for what you are trying to do. Can you explain a bit more what you want to do in what conditions and maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure, so it can continue on after the bridge unless you specify otherwise. Mike On Dec 7, 2009, at 1:31 PM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > > data="sofia/external/219 at 10.11.12.243,sofia/external/223 at 10.11.12.234,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245"/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. From mrene_lists at avgs.ca Tue Dec 8 08:11:40 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Dec 2009 11:11:40 -0500 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: set hangup_after_bridge=true Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 7-Dec-09, at 1:31 PM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > data="continue_on_fail=NO_ANSWER,USER_BUSY"/> > data="sofia/external/219 at 10.11.12.243,sofia/external/ > 223 at 10.11.12.234,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245 > "/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not > fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Tue Dec 8 08:12:26 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:12:26 -0500 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> <191c3a030912071139t2a261e07g9b449bade1a092de@mail.gmail.com> Message-ID: <19B11DE0-0F4A-4641-9B53-D2ED21261D48@jerris.com> If this issue continues after another update and re bootstrap/configure, please open up a bug on jira.freeswitch.org under build system, assign to me, and attach the config.log and config.status file from the root of your freeswitch src dir. Mike On Dec 7, 2009, at 2:39 PM, Anthony Minessale wrote: > try rerunning the ./bootstrap.sh > > > On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards wrote: > When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Monday, December 07, 2009 7:44 AM > To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. > > Best Regards, > Jerry > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Saturday, December 05, 2009 7:30 PM > To: Jerry Richards > Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > >> Why are you changing the 3pcc setting, is this an invite with no sdp? >> you need to take a trace from FS. >> >> 1) update to latest trunk first so line number match up. >> 2) issue these commands >> >> sofia profile internal siptrace on >> console loglevel debug >> >> save the output and put it on pastebin http://pastebin.freeswitch.org >> >> >> >> >> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9c5b118f/attachment-0002.html From mike at jerris.com Tue Dec 8 08:14:29 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:14:29 -0500 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: References: Message-ID: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> Please re-test this with svn trunk of freeswitch and if it is still the case open up a bug on jira.freeswitch.org in the build system catagory assigned to me and attach the config.log and config.status from the libs/esl dir to the bug. Mike On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote: > Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Monday, December 07, 2009 1:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 > > The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. > > -SOLINK=-shared -Xlinker -x > +SOLINK=-dynamiclib -Xlinker -x > > > Thats all you usually fix for the mac. > > > /b > > > > On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: > > > I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can?t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation -> MAC os X. > I have also googled this, and don?t see what I am doing wrong. Anybody there that can help? > applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install > make MYLIB="../libesl.a" SOLINK="-Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php > g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. > /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: > _main > __convert_to_string > __efree > __emalloc > __estrndup > __zend_get_parameters_array_ex > __zend_list_find > __zval_copy_ctor > _compiler_globals > _convert_to_long > _zend_error > _zend_get_constant > _zend_hash_find > _zend_register_list_destructors_ex > _zend_register_long_constant > _zend_register_resource > _zend_rsrc_list_get_rsrc_type > _zend_wrong_param_count > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make: *** [phpmod] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/2e15d6b1/attachment-0002.html From rob4manhere at gmail.com Tue Dec 8 08:14:59 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Tue, 8 Dec 2009 10:14:59 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Message-ID: I'm using FreeSWITCH in front of Asterisk without any issue. Stick with the latest trunk. Can you set your loglevel to debug and pastebin your log? Here are some additional tips to help us help you :) http://wiki.freeswitch.org/wiki/Reporting_Bugs Rob On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/83a2bf78/attachment-0002.html From mike at jerris.com Tue Dec 8 08:16:05 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:16:05 -0500 Subject: [Freeswitch-users] no hangup on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > Sorry no, apart from the fact that I was seeing the hangup. > > > I?m wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for ?*? and force a hangup? I don?t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: > Hi all, > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I?m not seeing a hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it?s not being fired. Does anyone have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d0816c10/attachment-0002.html From mike at jerris.com Tue Dec 8 08:19:42 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:19:42 -0500 Subject: [Freeswitch-users] compilation error of skypiax_protocol.c In-Reply-To: <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> References: <13529f9d0912022202g3254517dscc635bfb9c8b4439@mail.gmail.com> <9387ED56-4CDE-4685-AFF7-98E930BBA1F9@avgs.ca> <13529f9d0912022233i368b84c7s9bd960eb1380b53e@mail.gmail.com> <9CB60375-4E27-44CE-AE01-FE79BCF80D79@avgs.ca> <13529f9d0912022249p71c2a56awe0a57ee6ce5e77a4@mail.gmail.com> <13529f9d0912030024h76176df3j4848a76a0b85d9d6@mail.gmail.com> <13529f9d0912080109x4eb12ee6j53a3c49bc85a7106@mail.gmail.com> Message-ID: If you can off list provide me with remote login information to this box I can troubleshot the issue. Mike On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote: > Hi Jo?o, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. > > Here's how I normally update FreeSwitch: make clean && svn up && ./bootstrap.sh && ./configure && make install > > If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. > > Regards, > -Jingwei > > > 2009/12/8 Jo?o Mesquita > Maybe, just maybe isse that make target to reconf libtiff? > > Regards, > > JM > > > On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang wrote: > I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: > > gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o > at_interpreter.c: In function ???command_search???: > at_interpreter.c:5299: error: ???COMMAND_TRIE_LEN??? undeclared (first use in this function) > at_interpreter.c:5299: error: (Each undeclared identifier is reported only once > at_interpreter.c:5299: error: for each function it appears in.) > at_interpreter.c:5308: error: ???command_trie??? undeclared (first use in this function) > at_interpreter.c: In function ???at_interpreter???: > at_interpreter.c:5424: error: ???at_commands??? undeclared (first use in this function) > make[8]: *** [at_interpreter.lo] Error 1 > > make[7]: *** [all] Error 2 > make[6]: *** [all-recursive] Error 1 > make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 > make[4]: *** [install] Error 1 > make[3]: *** [mod_voipcodecs-install] Error 1 > make[2]: *** [install-recursive] Error 1 > > However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. > > Regards, > -Jingwei > > > > On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang wrote: > No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. > > Regards, > -Jingwei > > > On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene wrote: > Hi, > > That one is on your side. If you changed/updated system libs it might be worth doing another ./configure > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: > >> Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. >> >> gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc >> ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory >> make[8]: *** [at_interpreter_dictionary.h] Error 127 >> make[7]: *** [all] Error 2 >> make[6]: *** [all-recursive] Error 1 >> make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 >> make[4]: *** [install] Error 1 >> make[3]: *** [mod_voipcodecs-install] Error 1 >> make[2]: *** [install-recursive] Error 1 >> >> Do you have idea about this one? >> >> Thanks! >> >> On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene wrote: >> Consider it fixed. >> Committed revision 15765. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: >> >>> Hi Guys, >>> >>> I got a compilation error of skypiax_protocol.c with the latest version r15764. >>> >>> Compiling skypiax_protocol.c... >>> cc1: warnings being treated as errors >>> skypiax_protocol.c: In function ???X11_errors_handler???: >>> skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c: In function ???skypiax_send_message???: >>> skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c: In function ???skypiax_do_skypeapi_thread_func???: >>> skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code >>> skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code >>> make[5]: *** [skypiax_protocol.o] Error 1 >>> make[4]: *** [install] Error 1 >>> make[3]: *** [mod_skypiax-install] Error 1 >>> make[2]: *** [install-recursive] Error 1 >>> >>> I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d424e43e/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 8 08:22:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 10:22:03 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> Message-ID: <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> We could check it out for you if you want to contact me and give me ssh access. Or I can provide the instructions get it into the 100% cpu usage state then do the following without stopping FS. 1) run top -H and sort so all the FS threads are at the top and screen cap it so we can see which thread id is using the most cpu. 2) make sure you have gdb installed and issue this command from the build root ./support-d/fscore_pb gcore cpu_race_issue then we can compare the thread using the most cpu with the trace and locate your problem. On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ede6c2b8/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 8 08:28:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 10:28:04 -0600 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel wrote: > I got the combination Lua with direct access to the core Sqlite database > to work. Hurray, maybe I?m not as stupid as A.M II hints? > > The problem was that Lua did not ?like?: > > > > require "luasql.sqlite" > > env = luasql.sqlite() > > con = assert(env:connect("/usr/local/freeswitch/db/core.db")) > > > > After changing it to > > > > require "luasql.sqlite3" > > env = luasql.sqlite3() > > con = assert(env:connect("/usr/local/freeswitch/db/core.db")) > > > > And seeing that there was a symlink in one of the right directories called > with the name: sqlite3.so, it worked. > > > > Changing the core db into a MySQL via ODBC caused some problems even after > it seemed to work. For instance, console help caused an error with an error > description indicating that a SQL SELECT query including the reserved word > key has been fired. > > > > It this problem likely to be solved if I used another version of the MySQL? > > > > *Jon Br?el* > Consiglia Telecommunications > > DK-2960 Rungsted Kyst > Tel: +45 45 16 1000 > Mob: +45 26 15 30 60 > > CVR: 27047882 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/74c4d067/attachment-0002.html From mike at jerris.com Tue Dec 8 08:40:40 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:40:40 -0500 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> References: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> Message-ID: <3D5FFA01-DCA1-48E4-970C-8635B2F5F50E@jerris.com> I changed the name of key to ikey in trunk. Mike > Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. > > > It this problem likely to be solved if I used another version of the MySQL? > > > Jon Br?el > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/ca6d85a1/attachment-0002.html From mike at jerris.com Tue Dec 8 08:42:38 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 11:42:38 -0500 Subject: [Freeswitch-users] Force presence status manually In-Reply-To: <4B1E7894.2050101@gmx.net> References: <4B1E7894.2050101@gmx.net> Message-ID: The best way to solve this is probably to share the db for presence and registration between those boxes. If you take a look at the default configs the settings should be commented there. Mike On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote: > Hello, > > is there a way to manually force a presence status update? > In our scenario we have a Freeswitch cluster. As phones sometimes > register on one and one time on another machine via the load balancer, > we cannot dial via user/exten. Instead we dial each phone by it's > register string via xml-curl. That way -when a phone is called - other > phones who subscribed to this phone, do not receive a message to update > their presence status. > Is there a way to force the pesence status of a phone manually in the > dialplan? > We may then set the status before bridging and then reset it with a > hangup hook. > From msc at freeswitch.org Tue Dec 8 09:39:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Dec 2009 09:39:16 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: Message-ID: <87f2f3b90912080939s473feda4n7ea21fc995905b25@mail.gmail.com> On Tue, Dec 8, 2009 at 3:46 AM, Jon Bruel wrote: > I got the combination Lua with direct access to the core Sqlite database > to work. Hurray, maybe I?m not as stupid as A.M II hints? > Tsk tsk! He didn't actually hint that you were "stupid" - all he said was that doing ODBC and configuring databases isn't something as simple as flipping on a switch. It takes a bit of knowledge, much of which is hard-earned through experience. Trying to get it all up and running by emailing the list every time something goes wrong is like trying to learning how to change the oil in your car and emailing the Audi-users list every time something doesn't go as expected: yeah, you can probably learn something, possibly you can get it working, but it's grossly inefficient. You'd be much better off paying someone a few euros to come out and give you a lesson because in the long run it would save you both time and money. Just my $0.02. (Don't know what that is in euros...) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c3c40e9c/attachment-0002.html From abeka at greatiam.com Tue Dec 8 09:47:53 2009 From: abeka at greatiam.com (Otis) Date: Tue, 08 Dec 2009 17:47:53 +0000 Subject: [Freeswitch-users] Mutual Registration of servers In-Reply-To: <4B1D4E92.1040204@greatiam.com> References: <4B1CAF25.6010706@greatiam.com> <87f2f3b90912070943p5d41b9f3na76e8d390b0de5af@mail.gmail.com> <4B1D4E92.1040204@greatiam.com> Message-ID: <4B1E9149.80302@greatiam.com> Otis wrote: >
Michael > Collins wrote: >> >> >> On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah >> > wrote: >> >> Pardon me if this has been addressed already. >> How does one go about having in the simplest instance 2 servers >> registering with each other on startup whereby the users registering >> would be able to call each other. >> The 2 servers are in different domains. >> >> Thanks. >> >> >> Are the two servers in different locations? Different LANs? Is NAT >> involved? Just checking. Really this is just a matter of loading the >> default config on each machine and then making some decisions about >> the dialplan: do you want prefix dialing so that you can have ext >> 1000 at both locations or do you want to have something like >> 1000~1099 at location A and 1100~1199 at location B? From there it's >> just a matter of creating the gateways on each machine and adding a >> dialplan entry to handle the routing. >> -MC >> > Hello Michael > Thanks > Are the two servers in different locations? Yes > Different LANs? Yes > Is NAT involved? Yes but for my test Nat is not . The production setup > I have in mind will certainly have Nat > Each location will have their won set of extension but there could be > some overlap. > On server A a user would dial,. for example, 98 followed by the > extension number of the user on server B and the call would then be > routed to the extension on server B. And the same could be from > Server B to a user on Server A > > MC > > Thanks > > . > > >
> Please olks could someone let meknow if it is at possible. I have tried using the connecting to Asterisk without success, mimicked the link to a gateway unsuccessfully. Could someone please let me kno which .xml files to create etc. Thanks From Russell.Mosemann at cune.org Tue Dec 8 08:38:07 2009 From: Russell.Mosemann at cune.org (Russell.Mosemann at cune.org) Date: Tue, 8 Dec 2009 16:38:07 -0000 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: Message-ID: <20091208163807.DC92C45048A@mail.cune.org> Brian West said: > The fun part comes when you try to link that 32bit .a file into a > 64bit so file. That would require a dual-core processor. One core would be 32 bit and the other core would be 64 bit. ;-) -- Russell Mosemann ________________________________________________________ Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! From brian at freeswitch.org Tue Dec 8 10:00:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 12:00:05 -0600 Subject: [Freeswitch-users] Skype SIP Beta In-Reply-To: <20091208163807.DC92C45048A@mail.cune.org> References: <20091208163807.DC92C45048A@mail.cune.org> Message-ID: Well the fun part is you can't link them. :P /b On Dec 8, 2009, at 10:38 AM, wrote: > That would require a dual-core processor. One core would be 32 bit and > the other core would be 64 bit. ;-) > > -- > Russell Mosemann From jbr at consiglia.dk Tue Dec 8 10:00:54 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 8 Dec 2009 19:00:54 +0100 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> References: <191c3a030912080828y6ac8e20erf95a9ed2c8d6f08e@mail.gmail.com> Message-ID: Point taken Anthony. Naturally you are not going to work for me for free. But I'm a bit confused about the statement that "I'm rude". That's not my purpose to be. And I certainly do hope that this is not just a question of a cultural clash between an elderly man with a Phd in black holes from a European background and a young American FS genius. But frankly, I did believe that focus regarding changes and new developments was somewhat guided by the input you get from the users list, including changes which makes the FS easier to access for newbies, but maybe I'm wrong. That's my last comment, hope we can continue the exchange of views in a good spirit. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 8. december 2009 17:28 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel > wrote: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not "like": require "luasql.sqlite" env = luasql.sqlite() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) After changing it to require "luasql.sqlite3" env = luasql.sqlite3() con = assert(env:connect("/usr/local/freeswitch/db/core.db")) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/7b83ab6d/attachment-0002.html From xengelpublicx at gmail.com Tue Dec 8 10:14:02 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Tue, 08 Dec 2009 21:14:02 +0300 Subject: [Freeswitch-users] presense. freeswitch vs. spa962+932 Message-ID: <4B1E976A.5080207@gmail.com> Hello, I tune in the presence freeswitch and linksys spa962 932 I had a few qusteions: 1) if $ PROXY specified domain name and not ip the phone records. But all the buttons on spa932 blinking orange indicating that no subscriptions. phone logs like this: Call-ID: 76e0f816-9617ab46 at 192.168.0.100 User: 100 at 192.168.50.10 Contact: "user" Agent: Linksys/SPA962-6.1.3 (a) Status: Registered (UDP-NAT) (unknown) EXP (2009-12-08 21:26:14) Host: pbx0.test.lan IP: 192.168.0.100 Port: 1024 Auth-User: 100 Auth-Realm: pbx0.test.lan MWI-Account: 100 at 192.168.50.10 while the phone is not a nat. spa932 shows that subscriptions present. 2) how to see that now there is a basis of presence of fs_cli? 3) Can I configure two fs a mutually shared presence? This is done using ? Thabks. From spencer at 5ninesolutions.com Tue Dec 8 10:14:42 2009 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 8 Dec 2009 10:14:42 -0800 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> Message-ID: Hmm.. It doesn't seem to be a problem with Asterisk < 1.6.0.13. Asterisk 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this after an upgrade to 1.6.0.19. We're using xen on all our machines with 250hz timers. Could that be a problem? When I get a change I'll try to recreate this with a few more virtual machines to try to debug it. Spencer On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote: > We could check it out for you if you want to contact me and give me > ssh access. > Or I can provide the instructions > > get it into the 100% cpu usage state then do the following without > stopping FS. > > 1) run top -H and sort so all the FS threads are at the top and > screen cap it so we can see which thread id is using the most cpu. > 2) make sure you have gdb installed and issue this command from the > build root > ./support-d/fscore_pb gcore cpu_race_issue > > then we can compare the thread using the most cpu with the trace and > locate your problem. > > > > On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason > wrote: > Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. > The Asterisk boxes are individual hosted PBXs but they are configured > with identical software. This a x86_64 CentOS 5.4 system. I've tried > 1.0.4 and the latest svn with the same results. Basically Freeswitch > registers with outbound providers and I can send and receive test > calls. Then without warning, i.e. the Asterisk boxes are all idle and > there are no calls, the Freeswitch process starts using 100% of the > cpu. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/b127e2ae/attachment-0002.html From msc at freeswitch.org Tue Dec 8 10:19:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Dec 2009 10:19:39 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here! Message-ID: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> Greetings, The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH 1.0.5 pre-release version. Please check out the release announcement. Let's all get updated as soon as possible. Also, please report bugs right away and follow up when the developers need further information. We have had to close out some bugs due to lack of information from the one reporting. Of course, those running SVN trunk are asked to do a "make current" as soon as reasonably possible. The devs love it when you are on the latest trunk. :) Thanks again for all of your help! Let's keep up the good work and we'll have 1.0.5 available in no time. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/1e42be41/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 8 10:23:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 12:23:39 -0600 Subject: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers In-Reply-To: References: <3F991B2D-DC11-488A-9AC5-EE55BC5B79AA@5ninesolutions.com> <191c3a030912080822h7050432dj3e0a1a8785c3cf03@mail.gmail.com> Message-ID: <191c3a030912081023tdbdf0efjfd474b53214ca930@mail.gmail.com> would not be able to even guess without some data to examine. On Tue, Dec 8, 2009 at 12:14 PM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > Hmm.. It doesn't seem to be a problem with Asterisk < 1.6.0.13. Asterisk > 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this > after an upgrade to 1.6.0.19. We're using xen on all our machines with > 250hz timers. Could that be a problem? When I get a change I'll try to > recreate this with a few more virtual machines to try to debug it. > > Spencer > > On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote: > > We could check it out for you if you want to contact me and give me ssh > access. > Or I can provide the instructions > > get it into the 100% cpu usage state then do the following without stopping > FS. > > 1) run top -H and sort so all the FS threads are at the top and screen cap > it so we can see which thread id is using the most cpu. > 2) make sure you have gdb installed and issue this command from the build > root > ./support-d/fscore_pb gcore cpu_race_issue > > then we can compare the thread using the most cpu with the trace and locate > your problem. > > > > On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason < > spencer at 5ninesolutions.com> wrote: > >> Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. >> The Asterisk boxes are individual hosted PBXs but they are configured >> with identical software. This a x86_64 CentOS 5.4 system. I've tried >> 1.0.4 and the latest svn with the same results. Basically Freeswitch >> registers with outbound providers and I can send and receive test >> calls. Then without warning, i.e. the Asterisk boxes are all idle and >> there are no calls, the Freeswitch process starts using 100% of the cpu. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/e2e659f6/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 8 10:25:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 12:25:52 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here! In-Reply-To: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> References: <87f2f3b90912081019v5f481b45mf727ec6db339ef96@mail.gmail.com> Message-ID: <191c3a030912081025l1481cda9rc7cbbd0343ef51cc@mail.gmail.com> Let's see if we can beat Duke Nukem Forever! On Tue, Dec 8, 2009 at 12:19 PM, Michael Collins wrote: > Greetings, > > The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH > 1.0.5 pre-release version. Please check out the release announcement. > Let's all get updated as soon as possible. Also, please report bugs right > away and follow up when the developers need further information. We have had > to close out some bugs due to lack of information from the one reporting. > > Of course, those running SVN trunk are asked to do a "make current" as soon > as reasonably possible. The devs love it when you are on the latest trunk. > :) > > Thanks again for all of your help! Let's keep up the good work and we'll > have 1.0.5 available in no time. > > -Michael > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f267b1bf/attachment-0002.html From JCasale at activenetwerx.com Tue Dec 8 10:26:58 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 8 Dec 2009 18:26:58 +0000 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <26694801.post@talk.nabble.com> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> Message-ID: >For those interested, here's how to compile and install Dahdi (which doesn't >need Asterisk at all, unlike some docs on the Net seem to imply due to >references to /etc/asterisk/*.conf): I understand that Some Debian based distro's have Dahdi in their repo's making it simple, but not many know that Digium runs its own repo for rpm based distros: http://packages.asterisk.org/ Can't get easier than that... jlc From kristian.kielhofner at gmail.com Tue Dec 8 10:30:12 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 8 Dec 2009 13:30:12 -0500 Subject: [Freeswitch-users] Choppy sound with PCMU In-Reply-To: <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> References: <26594250.post@talk.nabble.com> <191c3a030912031010x1ac0a766o6509550bf114332f@mail.gmail.com> <26630994.post@talk.nabble.com> <191c3a030912031240v39ec7888oc4b0a6cec63d1d6d@mail.gmail.com> <26633739.post@talk.nabble.com> <191c3a030912031549tdc6057dh294e678f1c5b72a5@mail.gmail.com> <4B1963E8.7050204@cartissolutions.com> <191c3a030912041148g76cec941v50417178bde62917@mail.gmail.com> <26678873.post@talk.nabble.com> <191c3a030912070800q5f318bads1ec60dc7520671c1@mail.gmail.com> Message-ID: <2d9149cd0912081030n14729f1en1ff0ec1a1506357@mail.gmail.com> For reference, here is the AstLinux kernel config for the ALIX: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/target/device/alix/linux.config?view=markup We've got what I consider to be excellent support for the ALIX - most of the developers use them and they are very popular in the community. On Mon, Dec 7, 2009 at 11:00 AM, Anthony Minessale wrote: > Did you do each thing alone too to tell the difference? > -hp alone, disable monotonic alone (i did not see you mention the disable > monotonic) > > as for your 4ms thing, yes we require high resolution timing, if we ask to > sleep 1000 microseconds that is what we need it to sleep for or at least as > close as possible, and the main reason that thread is never sleeping is > because you can't actually count on it to run every 1ms but you mostly can. > Hence the whole philosophy on only making 1 thread run hot all the time to > ensure that the rest don't have to repeat the same algorithm.? We focus on > high end performance this was the point of your experimentation because we > will need to use a compile time defines and other logic to make it more > efficient on your platform, a platform which we are not using.? I am curious > what would happen if you install Kristian's astlinux on one of your devices, > i think you should also compare the kernel versions. > > > What OS are you running anyway? > > Here are some more things to try (running plain trunk with no mods) do these > systematically each alone and all together with/without -hp or disable > monotonic etc to see what different combos create > > comment out this line (line 10) > #define DISABLE_1MS_COND > > rebuild, this tells it to run a conditional at 1ms in the same timer thread > which will make all the switch_cond_next share a 1ms conditional instead of > doing microsleeps > > next > > some kernels/devices work better using select(0) for sleep where others work > better using usleep. > comment out line 109 > apr_sleep(t); > > and try > usleep(t) > > also mac works better using nanosleep so you could try changing it so it > uses the code starting at 101 instead. > > > also your claim about JS should be investigated because I do not think it > should be the case. > but you may want to move this to a jira http://jira.freeswitch.org > > As for the asterisk comparison, > not sure how to answer you, that's your decision. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From testa at voicetechnology.com.br Tue Dec 8 10:41:15 2009 From: testa at voicetechnology.com.br (Fernando Gregianin Testa) Date: Tue, 8 Dec 2009 16:41:15 -0200 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones Message-ID: Dear list, Some Nec phones sends DTMF RFC2833 with payload 101 during the call, but have negotiated a different one on SDP. When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 we notice this phone sends the following INVITE packet and RTP packets: http://pastebin.freeswitch.org/11433 Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap Is there any parameter to tweak FS in such a way to force understand 101 packets as DTMF? Thank you in advance! Fernando Testa PS: On pcap you have the following IPs: FS at 10.91.10.210 Nec Pbx 10.91.10.22 phone 10.91.10.85 From brian at freeswitch.org Tue Dec 8 10:51:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Dec 2009 12:51:32 -0600 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones In-Reply-To: References: Message-ID: Best option for you is to use 96 in the sofia profile you're using to talk to these broken devices. /b On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: > Dear list, > > Some Nec phones sends DTMF RFC2833 with payload 101 during the call, > but have negotiated a different one on SDP. > When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 > we notice this phone sends the following INVITE packet and RTP > packets: http://pastebin.freeswitch.org/11433 > Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap > > Is there any parameter to tweak FS in such a way to force understand > 101 packets as DTMF? > Thank you in advance! > > Fernando Testa > PS: On pcap you have the following IPs: > FS at 10.91.10.210 > Nec Pbx 10.91.10.22 > phone 10.91.10.85 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From ken at ksac.com Tue Dec 8 10:56:36 2009 From: ken at ksac.com (Kendall Stauffer) Date: Tue, 8 Dec 2009 10:56:36 -0800 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> References: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> Message-ID: Hey you guys, I know this isn't the right place for this, but I have been working with asterisk for 5 years now, and just got freeswitch working (on windows, not os x yet). All I can say is AWESOME --- thanks so much!!!! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 08, 2009 9:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/d3af9e8f/attachment-0002.html From mailinglist at fribert.dk Tue Dec 8 11:20:29 2009 From: mailinglist at fribert.dk (mailinglist) Date: Tue, 08 Dec 2009 20:20:29 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1DFABC020000E1000002C2@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> Message-ID: <4B1EB50D020000E1000002C7@mail.fribert.dk> Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f925530e/attachment-0002.html From mike at jerris.com Tue Dec 8 11:57:28 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Dec 2009 14:57:28 -0500 Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> Message-ID: <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> Our plan for 1.0.5 is that we will also have rpm and deb packages for many distros on our own repo. Stay tuned. This has been another major reason for the delay in 1.0.5. Mike On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote: >> For those interested, here's how to compile and install Dahdi (which doesn't >> need Asterisk at all, unlike some docs on the Net seem to imply due to >> references to /etc/asterisk/*.conf): > > I understand that Some Debian based distro's have Dahdi in their repo's making it > simple, but not many know that Digium runs its own repo for rpm based distros: > > http://packages.asterisk.org/ > > Can't get easier than that... > > jlc > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jerry.richards at teotech.com Tue Dec 8 12:35:01 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 Dec 2009 12:35:01 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/81bdc073/attachment-0002.html From jerry.richards at teotech.com Tue Dec 8 12:57:39 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 Dec 2009 12:57:39 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/9755354a/attachment-0002.html From larclap at yahoo.com Tue Dec 8 13:09:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 8 Dec 2009 13:09:03 -0800 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> <253F208905D14A28B8334C913CAEBFCC@greyhawk.tonecommander.com> Message-ID: <011401ca784a$ac065690$041303b0$@com> Can you copy the address of the pastebin so that people can see it? After you hit the Send button, the address is posted back at the top of your browser, like: http://pastebin.freeswitch.org/11441 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _____ From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/c7eef752/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 8 13:21:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 15:21:14 -0600 Subject: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP In-Reply-To: References: <1DB1AEF56E094F94A731065F2F2FB091@greyhawk.tonecommander.com> <191c3a030912041259k5adbad5cy8821e2abe7127cd2@mail.gmail.com> <6424DB76-803D-4AE9-A201-334B7B8286D1@jerris.com> Message-ID: <191c3a030912081321j5c875e67r5a0ef6f6d018fa65@mail.gmail.com> are you using more than one profile here? if so you have to repeat the siptrace on command for each one. This trace makes little sense to me because I think half of it is missing. but you can see several packets coming in like 20 times each which means you have some kind of nat or network problem causing the other end of this call to send retries on all the packets. On Tue, Dec 8, 2009 at 2:57 PM, Jerry Richards wrote: > Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 > > Thanks, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Tuesday, December 08, 2009 12:35 PM > > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Anthony and Michael, > > I downloaded the latest trunk, rebuilt it, and re-ran the test with the > logs that Anthony told me to turn on. I put the results up in the PasteBin. > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 10:49 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > When I got the latest trunk the make gets an error. Should I perhaps > disable the mod_amr? > > making all mod_amr > make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. > Stop > > The method I used to get the latest trunk follows: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch > > Best Regards, > Jerry > > ------------------------------ > *From:* Jerry Richards [mailto:jerry.richards at teotech.com] > *Sent:* Monday, December 07, 2009 7:44 AM > *To:* 'Michael Jerris'; 'freeswitch-users at lists.freeswitch.org' > *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > I am changing the 3pcc setting because one of my gateways sends INVITEs > without SDP. I will try to update to the latest trunk today and capture > traces as Anthony described. If I can't do it today, it might be at the end > of the week. > > Best Regards, > Jerry > > > ------------------------------ > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Saturday, December 05, 2009 7:30 PM > *To:* Jerry Richards > *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION > UNREACHABLE When Gateway Sends RTP > > Jerry- > > Any update on this? > > Mike > > On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: > > Why are you changing the 3pcc setting, is this an invite with no sdp? > you need to take a trace from FS. > > 1) update to latest trunk first so line number match up. > 2) issue these commands > > sofia profile internal siptrace on > console loglevel debug > > save the output and put it on pastebin http://pastebin.freeswitch.org > > > > > On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards > wrote: > >> >> I have Mediant 1000 gateway, and for some reason, when I make an outbound >> call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A >> Wireshark trace shows that FS is replying to the gateway's inbound RTP >> packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP >> packets to the same port that FS specified in the outbound INVITE. It >> appears in the log that FS is discarding the 200 OK from the gateway. >> >> I disabled the Firewall and SELinux on the Freeswitch machine. I tried >> changing to "true" and also "proxy", but it has no effect. >> >> Anyone know what could be the issue? I posted the Freeswitch log in the >> pastebin. >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/660ad668/attachment-0002.html From orien at tx.rr.com Tue Dec 8 13:25:22 2009 From: orien at tx.rr.com (Orien Love) Date: Tue, 08 Dec 2009 15:25:22 -0600 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: References: Message-ID: <4B1EC442.7010603@tx.rr.com> I am looking for a 4 port FXO card to use with my PfSense installation of freeswitch. does anybody know if the Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? or could somebody recommend one that would. Thank You Orien From anthony.minessale at gmail.com Tue Dec 8 13:33:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Dec 2009 15:33:30 -0600 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: <4B1EC442.7010603@tx.rr.com> References: <4B1EC442.7010603@tx.rr.com> Message-ID: <191c3a030912081333t23c27bcbh720fa5930a413ff@mail.gmail.com> I dont think there are any supported hw for bsd, there are legacy sangoma and zaptel drivers floating around but they are not supported by the vendors. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love wrote: > I am looking for a 4 port FXO card to use with my PfSense installation > of freeswitch. does anybody know if the > Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > or could somebody recommend one that would. > > Thank You > Orien > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/1242b553/attachment-0002.html From rsavage at KingBallow.com Tue Dec 8 13:36:23 2009 From: rsavage at KingBallow.com (Reece Savage) Date: Tue, 8 Dec 2009 15:36:23 -0600 Subject: [Freeswitch-users] Rhino Cards for sale R2T1-EC and R24FXX-EC In-Reply-To: References: <26694069.post@talk.nabble.com><20091208141536.8AD6E421BDE@mail.cune.org><26694801.post@talk.nabble.com> Message-ID: <8E4ACA7747F7F641991455BC157390C801450356@srv-nash-ex.mail.kingballow.com> I have 2 Rhino cards for sale if anyone needs one. They are both Best Offer. I have a R2T1-EC and a R24FXX-EC with 12 dual FXS modules. Both have never been used more than a few times for testing purposes. Both cards work fine and are guaranteed not to be DOA. Reece Savage Information Technology Manager King & Ballow Law Offices 315 Union Street Suite 1100 Nashville, TN? 37201 Phone (615) 726-5525 Fax (615) 254-7907 rsavage at kingballow.com From nandy1925 at gmail.com Tue Dec 8 14:45:15 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 9 Dec 2009 06:45:15 +0800 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1EB50D020000E1000002C7@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> Message-ID: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: > Hi All > > Ok, after reading a bit more I think I see what I've done wrong, but I > don't know how to fix it properly. > Looking in the Dialplan directory I see the following: > default (dir) > default.xml > features.xml > public (dir) > public.xml > > Under the default dir the webinterface has created the 001_musimi.dk.xml > file that I've created. > But as I understand it, it doesn't use it. > > How do I make it use it, I would very much like to keep the webinterface > editor, and not have to do it via ssh and vi all the time. > > >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i > meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: > Hi Mark > > Ok, thanks. > Yes I have a gateway placed in external called musimi.dk (or should it be > in public?), and I'll just create the empty XML's in lan to get rid of that > error. > > I'll remove the second part of the dialplan, my idea was that it was needed > for calls between sip phones hooked up to the freeswitch. > > Now the remaining problem: > When I call ext 1002 from ext 1001 I see this message and get an error, the > same goes for dialing 0XXXXXXXX to get an external number: > > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing > 1001->1002 in context default > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 > [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. > Cause: NO_ROUTE_DESTINATION > 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup > sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 > (sofia/external/$1) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/external/$1 [CS_DESTROY] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( > sofia/internal/1001 at 10.11.12.25) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] > I don't see any mention of the statements in the Dialplan, so for me it > looks like it haven't registered the Dialplan? > > Best regards > Kenneth > > >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen > 659603.29094.qm at web56408.mail.re3.yahoo.com: > > Question ---------------------------------------------- > If I do a reloadxml it gives me this output on the console: > freeswitch at firewall.fribert.dk> > reloadxml > 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > > I'm not sure if it's a genuine problem,as I can see it, it just complains > that I haven't created any sip_profiles in /lan, but is that necessary? > > Response: ---------------------------------------------- > This isn't really a problem. To get rid of the error simply put a blank xml > file into each folder as in the internal and external directories. Dump the > lan directory and lan profile as mentioned earlier. > > Question ---------------------------------------------- > > Extension Name musimi.dk > Enabled true > Order 001 > Description ... > > condition ^0(.\d+)$ > action bridge sofia/gateway/musimi.dk/$1 > > Response: ---------------------------------------------- > > This is correct as long as you have a gateway that is registered called > musimi.dk > > Question ---------------------------------------------- > Extension Name 10.11.12.25 > Enabled true > Order 002 > Description ... > > action bridge sofia/internal/$ > > Response: ---------------------------------------------- > > No idea what this is for its not needed as far as I can tell. > > > Now please summarize what you still need help on. > > > Mark J Crane > http://fusionpbx.com > pfSense FreeSWITCH package developer > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/b0204313/attachment-0002.html From nandy1925 at gmail.com Tue Dec 8 14:50:58 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 9 Dec 2009 06:50:58 +0800 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <4B1D4A08.80507@gmx.net> References: <4B1D4A08.80507@gmx.net> Message-ID: <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> this action can be accomplished using Group Dialing (Sequential). this may not answer your problem but have you considered it? -nandy On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX wrote: > I have a Problem with continue_on_fail. > > > I have setup a hunt group > > data="sofia/external/219 at 10.11.12.243,sofia/external/223 at 10.11.12.234 > ,sofia/external/236 at 10.11.12.188,sofia/external/101 at 10.11.12.245"/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/9a7a9515/attachment-0002.html From timuckun at gmail.com Tue Dec 8 16:24:11 2009 From: timuckun at gmail.com (Tim Uckun) Date: Wed, 9 Dec 2009 13:24:11 +1300 Subject: [Freeswitch-users] Lua and database access to core_db In-Reply-To: References: <8ADFFB51-6A0E-427D-AE76-2B98C4F3689D@freeswitch.org> Message-ID: <855e4dcf0912081624g114ccb47hb7bcfca957c4cc38@mail.gmail.com> On Wed, Dec 9, 2009 at 7:56 AM, Kendall Stauffer wrote: > Hey you guys, I know this isn?t the right place for this, but I have been > working with asterisk for 5 years now, and just got freeswitch working (on > windows, not os x yet). > > All I can say is AWESOME --- thanks so much!!!! Out of curiosity. Did you choose freeswitch because it runs on windows and asterisk doesn't? I find some people choose freeswitch because they don't know or want to use linux (obviously this doesn't apply to you) and some people choose it because they want a windows solution and asterisk doesn't run on windows. From ryannyl at gmail.com Tue Dec 8 19:21:13 2009 From: ryannyl at gmail.com (Ryanny Lin) Date: Wed, 9 Dec 2009 11:21:13 +0800 Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. Message-ID: <4bfcac7e0912081921m1f26d337we9e92b310ea2cd1e@mail.gmail.com> hi I don't know this driver for freebsd works or not, maybe you can check it out and try it. Here is the Digium's SVN repository: http://svn.digium.com/svn/dahdi/ It looks like driver for freebsd is under testing. Or use zaptel driver ... Change to root, then cd /usr/ports/misc/zaptel make install && make clean uh..., maybe PfSense doesn't have /usr/ports ...you can just install zaptel. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love > wrote: > > >* I am looking for a 4 port FXO card to use with my PfSense installation > *>* of freeswitch. does anybody know if the > *>* Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > *>* or could somebody recommend one that would. > *>* > *>* Thank You > *>* Orien > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > *>* http://www.freeswitch.org > *>** > > -- Sincerely regards, Wen-Jen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/2cb28b50/attachment-0002.html From mctch at yahoo.com Wed Dec 9 01:26:49 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 01:26:49 -0800 (PST) Subject: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. In-Reply-To: <4bfcac7e0912081921m1f26d337we9e92b310ea2cd1e@mail.gmail.com> Message-ID: <446840.56954.qm@web56405.mail.re3.yahoo.com> You need to use a zaptel package that was compiled for the operating system that pfsense uses. For pfSense 1.2.3 try this package:? ssh to pfsense then run: press 8 for the command line access cd /tmp fetch http://portableusbapps.com/packages/config/freeswitch/zaptel-1.4.6_7.tgz pkg_add zaptel-1.4.6_7.tgz Also you will need to use the FreeSWITCH Dev package on pfsense which has the openzap module. FreeSWITCH dev package is not perfect that is why it is still marked as dev. Best Regards, Mark J Crane P.S. For those that don't know the I created the pfSense FreeSWITCH package wanted a name for the project and finally after a lot of thought came up with FusionPBX. Last night I stayed up all night to see if I could get FusionPBX to work on pfSense. I did get it so that is a step in the right direction to get the latest version working on pfSense again. FusionPBX uses PDO (PHP Data Objects) which is available in PHP5. pfSense 1.2.3 uses PHP4 and so it required getting both PHP4 and PHP5 working on the same machine setup in a way that they don't conflict which I was able to achieve. So new version is getting closer. --- On Tue, 12/8/09, Ryanny Lin wrote: From: Ryanny Lin Subject: Re: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch. To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 8:21 PM hi I don't know this driver for freebsd works or not, maybe you can check it out and try it. Here is the Digium's SVN repository: http://svn.digium.com/svn/dahdi/ It looks like driver for freebsd is under testing. Or use zaptel driver ... Change to root, then cd /usr/ports/misc/zaptel make install && make clean uh..., maybe PfSense doesn't have /usr/ports ...you can just install zaptel. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love wrote: > I am looking for a 4 port FXO card to use with my PfSense installation > of freeswitch. does anybody know if the > Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? > or could somebody recommend one that would. > > Thank You > Orien > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely regards, Wen-Jen -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/ae94c7ba/attachment-0002.html From mctch at yahoo.com Wed Dec 9 01:28:07 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 01:28:07 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <187489.95329.qm@web56408.mail.re3.yahoo.com> Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch at yahoo.com --- On Tue, 12/8/09, Nandy Dagondon wrote: From: Nandy Dagondon Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All ? Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml ? Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark ? Ok, thanks. Yes I have a gateway placed in external?called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan?to get rid of that error. ? I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. ? Now the remaining problem: When I call ext 1002 from?ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: ? 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed.? Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? ? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console:freeswitch at firewall.fribert.dk> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name? musimi.dk Enabled true Order 001 Description? ... ? condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... ? action bridge? sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/3eaa110b/attachment-0002.html From Prometheus001 at gmx.net Wed Dec 9 02:12:04 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 09 Dec 2009 11:12:04 +0100 Subject: [Freeswitch-users] continue_on_fail In-Reply-To: <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> References: <4B1D4A08.80507@gmx.net> <7d0bfd8c0912081450m5c4c54fds37f54d2a4a779af9@mail.gmail.com> Message-ID: <4B1F77F4.6030302@gmx.net> Hello Nandy, thanks for your hint, but it's a bit more than that. In our application which is handled via XML-Curl, the user can define it's forwards on a web interface. He can enter mixed local or external numbers which are called sequentially or in parallel. Best regards Peter Nandy Dagondon schrieb: > this action can be accomplished using Group Dialing (Sequential). this > may not answer your problem but have you considered it? > -nandy > > > On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX > wrote: > > I have a Problem with continue_on_fail. > > > I have setup a hunt group > data="continue_on_fail=NO_ANSWER,USER_BUSY"/> > data="sofia/external/219 at 10.11.12.243 > ,sofia/external/223 at 10.11.12.234 > ,sofia/external/236 at 10.11.12.188 > ,sofia/external/101 at 10.11.12.245 > "/> > > I want the fallback user to be called whenever none of the previously > called 3 gateway numbers picks up or if they are all busy. > Therefore continue_on_fail=NO_ANSWER,USER_BUSY > > The fallback user is called, however if any of the previously called > gateways picks up and then hangs up, the fallback user is called > afterwards. > Means: The fallback user is always called. > > I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would > not fire > the next bridge if it gets a NORMAL_CLEARING. > > Am I thinking wrongly about this? > > I have added > > and this works, but I would like to specify more in detail the > conditions when to follow the next hunt group entry. > > Best regards > Peter > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jonas.gauffin at gmail.com Wed Dec 9 02:25:36 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 9 Dec 2009 11:25:36 +0100 Subject: [Freeswitch-users] OT: Spa2102 and call transfer Message-ID: Hello, I can't get call transfer to work with a SPA2102 adapter. I don't think it has something to do with FS, but I'm hoping someone here could help me. I do not get a new line in the phone (by pressing the R button), all DTMF tones are sent as audio to the other connected phone. Anyone got it working? Thanks, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/60f8b949/attachment-0002.html From timuckun at gmail.com Wed Dec 9 02:26:41 2009 From: timuckun at gmail.com (Tim Uckun) Date: Wed, 9 Dec 2009 23:26:41 +1300 Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <987536.45831.qm@web37508.mail.mud.yahoo.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> Message-ID: <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> On Tue, Dec 8, 2009 at 5:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. ?Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > ?Any suggestion would be appreciated. ?Thank you. If you could explain how you are doing the load balancing it would be really helpful to me. I am trying to do the same thing. From devel at thom.fr.eu.org Wed Dec 9 03:01:22 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Wed, 09 Dec 2009 12:01:22 +0100 Subject: [Freeswitch-users] =?utf-8?q?CLIP_on_FXS_channels_with_mod=5Fopen?= =?utf-8?q?zap?= In-Reply-To: <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> Message-ID: <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> I'm still working on this issue, and decided to take a look at the openzap code. First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid. I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span) By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it. Is my asumption (feature not implemented) correct ? Fran?ois On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote: Did you also update your wanpipe drivers and rebuild openzap again after you upgraded it? On Wed, Dec 2, 2009 at 2:12 AM, Fran?ois Legal wrote: So I did some tests and still I can not see CLIP on a phone connected to an FXS port. Whether the call is bridged from SIP UA or from an incoming call on FXO port does not change anything. Whether the parameter enable-caller-id=true is present or not in openzap.conf.xml does not change anything too. On that subject, sangoma support team says it must be freeswitch as this feature is supported and has been tested working. However, the good point is that I did not experience cuts in my call bridged from FXS to FXO with that new release. Fran?ois On Tue, 1 Dec 2009 19:02:11 -0600, Anthony Minessale wrote: upgrading always helps *something* not sure. but that is where we have to start because we have changed that code alot. On Tue, Dec 1, 2009 at 2:37 AM, Fran?ois Legal wrote: Sure, I'll try that. I'm just building freeswitch-snapshot that I downloaded from files.freeswitch.org [4] I also experience, when bridging a call from an FXS to FXO the call is cut after a random time (this does not appear when bridging SIP to FXO). Might this upgrade fix this problem also ? Fran?ois On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote: can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, Fran?ois Legal wrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. Fran?ois _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [8] http://www.freeswitch.org [9] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [10] ClueCon http://www.cluecon.com/ [11] Twitter: http://twitter.com/FreeSWITCH_wire [12] AIM: anthm MSN:anthony_minessale at hotmail.com [13] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [14] IRC: irc.freenode.net [15] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [16] iax:guest at conference.freeswitch.org/888 [17] googletalk:conf+888 at conference.freeswitch.org [18] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [20] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [21] http://www.freeswitch.org [22] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [23] ClueCon http://www.cluecon.com/ [24] Twitter: http://twitter.com/FreeSWITCH_wire [25] AIM: anthm MSN:anthony_minessale at hotmail.com [26] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [27] IRC: irc.freenode.net [28] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [29] iax:guest at conference.freeswitch.org/888 [30] googletalk:conf+888 at conference.freeswitch.org [31] pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [32] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [33] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [34] http://www.freeswitch.org [35] -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ [36] ClueCon http://www.cluecon.com/ [37] Twitter: http://twitter.com/FreeSWITCH_wire [38] AIM: anthm MSN:anthony_minessale at hotmail.com [39] GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com [40] IRC: irc.freenode.net [41] #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org [42] iax:guest at conference.freeswitch.org/888 [43] googletalk:conf+888 at conference.freeswitch.org [44] pstn:213-799-1400 Links: ------ [1] mailto:devel at thom.fr.eu.org [2] mailto:anthony.minessale at gmail.com [3] mailto:devel at thom.fr.eu.org [4] http://files.freeswitch.org [5] mailto:devel at thom.fr.eu.org [6] mailto:FreeSWITCH-users at lists.freeswitch.org [7] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] http://lists.freeswitch.org/mailman/options/freeswitch-users [9] http://www.freeswitch.org [10] http://www.freeswitch.org/ [11] http://www.cluecon.com/ [12] http://twitter.com/FreeSWITCH_wire [13] mailto:MSN%3Aanthony_minessale at hotmail.com [14] mailto:PAYPAL%3Aanthony.minessale at gmail.com [15] http://irc.freenode.net [16] mailto:sip%3A888 at conference.freeswitch.org [17] http://iax:guest at conference.freeswitch.org/888 [18] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [19] mailto:FreeSWITCH-users at lists.freeswitch.org [20] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [21] http://lists.freeswitch.org/mailman/options/freeswitch-users [22] http://www.freeswitch.org [23] http://www.freeswitch.org/ [24] http://www.cluecon.com/ [25] http://twitter.com/FreeSWITCH_wire [26] mailto:MSN%3Aanthony_minessale at hotmail.com [27] mailto:PAYPAL%3Aanthony.minessale at gmail.com [28] http://irc.freenode.net [29] mailto:sip%3A888 at conference.freeswitch.org [30] http://iax:guest at conference.freeswitch.org/888 [31] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org [32] mailto:FreeSWITCH-users at lists.freeswitch.org [33] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] http://lists.freeswitch.org/mailman/options/freeswitch-users [35] http://www.freeswitch.org [36] http://www.freeswitch.org/ [37] http://www.cluecon.com/ [38] http://twitter.com/FreeSWITCH_wire [39] mailto:MSN%3Aanthony_minessale at hotmail.com [40] mailto:PAYPAL%3Aanthony.minessale at gmail.com [41] http://irc.freenode.net [42] mailto:sip%3A888 at conference.freeswitch.org [43] http://iax:guest at conference.freeswitch.org/888 [44] mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/c963a88f/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Dec 8 15:59:55 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 8 Dec 2009 23:59:55 -0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! Message-ID: Thought I'd send this little hurrah! As there seems to have been a lot of negativity on this list lately. >From my point of view, having looked at many solutions out there, FS is still number one with regards to flexibility and performance. I cannot imagine doing what I'm using FS for, with any other product. Yes it's frustrating at times, but this is largely down to a lack documentation/samples. So, if you have a solution to a problem, share it by adding an entry on the WIKI. Kudos to AM and all the other dev's, as someone said once 'Don't let the bastards grind you down' Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/f9b84f4e/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Dec 8 15:48:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 8 Dec 2009 23:48:58 -0000 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: No doubt, but that's a little difficult as this only happens occasionally and I have 200 calls going on at the time. It's needle in the haystack stuff. Here's what I know. I have an external process listening for DTMF events. If I detect '*' I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn't exist, so I now do a double kill on the associated leg which I get from the event. I do not get a 'doesn't exist' message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. The kluge clears the B leg hang issue, so the pressure's off for me, but when I get a few nano seconds, I'll look at the code to see if there's anything obvious. Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 08 December 2009 16:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091208/21743be3/attachment-0002.html From codecomplete at free.fr Wed Dec 9 03:34:24 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 9 Dec 2009 03:34:24 -0800 (PST) Subject: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk? In-Reply-To: <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> References: <26694069.post@talk.nabble.com> <20091208141536.8AD6E421BDE@mail.cune.org> <26694801.post@talk.nabble.com> <47C35816-CE7A-47AC-8DDB-092380CCA9E9@jerris.com> Message-ID: <26708848.post@talk.nabble.com> Michael Jerris wrote: > Our plan for 1.0.5 is that we will also have rpm and deb packages for many > distros on our own repo. Stay tuned. This has been another major reason > for the delay in 1.0.5. Great news. I also prefer to use packages whenever possible, so as to know what software is installed in a host, and have the package manager handle conflicts and missing dependencies. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26708848.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Dec 9 05:58:46 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Dec 2009 08:58:46 -0500 Subject: [Freeswitch-users] CLIP on FXS channels with mod_openzap In-Reply-To: <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> References: <191c3a030912011702v735d823lb2d74df66b249b1f@mail.gmail.com> <337f1acf5a4df69bd33b87bc4227c720@thom.fr.eu.org> <191c3a030912021117w17f2b02bk19350c79b8608431@mail.gmail.com> <91ba59fbacc8df9e4f9837ac70af9421@thom.fr.eu.org> Message-ID: <76C4CA2A-4569-4EAA-83CF-E0EEDFC18242@jerris.com> I recall implementing that back when we released openzap, it should be in there unless someone chopped it out for some reason. Look for "zap_channel_send_fsk_data" Mike On Dec 9, 2009, at 6:01 AM, Fran?ois Legal wrote: > I'm still working on this issue, and decided to take a look at the openzap code. > > First, I figured out that the parameter name for callerid is enable_callerid rather than enable-callerid. > > I also figured out that this parameter defaults to TRUE (which is coherent with the observed behaviour on my FXO span) > > > By further checking the code, I figured out that presenting the callerid on an FXS port might not be implemented yet. I could see the code for retrieving the callerid from FXO but nothing to send it. > > > Is my asumption (feature not implemented) correct ? > > > Fran?ois > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/2cf60bed/attachment-0002.html From mike at jerris.com Wed Dec 9 06:00:43 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Dec 2009 09:00:43 -0500 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> Message-ID: <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> src/switch_ivr_bridge.c This could just as well be a glare condition when the call is in process of tearing down. Mike On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote: > No doubt, but that?s a little difficult as this only happens occasionally and I have 200 calls going on at the time. It?s needle in the haystack stuff. > > Here?s what I know. > > I have an external process listening for DTMF events. If I detect ?*? I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn?t exist, so I now do a double kill on the associated leg which I get from the event. I do not get a ?doesn?t exist? message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. > > The kluge clears the B leg hang issue, so the pressure?s off for me, but when I get a few nano seconds, I?ll look at the code to see if there?s anything obvious. > > Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: 08 December 2009 16:16 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > We will really need debug logs and sip traces to be able to figure out what exactly is going on here. > > Mike > > On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: > > > Sorry no, apart from the fact that I was seeing the hangup. > > > I?m wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for ?*? and force a hangup? I don?t seem to able to see this tone on the B leg though. > > Regards, > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 07 December 2009 19:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] no hangup on B leg > > > > On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: > Hi all, > > I?ll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I?m not seeing a hangup of the b leg at all. > > FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it?s not being fired. Does anyone have an idea what might be causing this? > > Regards, > > Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/8260f359/attachment-0002.html From jonas.gauffin at gmail.com Wed Dec 9 06:01:14 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 9 Dec 2009 15:01:14 +0100 Subject: [Freeswitch-users] OT: Spa2102 and call transfer In-Reply-To: References: Message-ID: I have the same problem with a HandyTone 502 adapter. Anyone got any hints to get the flash button to work? On Wed, Dec 9, 2009 at 11:25 AM, Jonas Gauffin wrote: > Hello, > > I can't get call transfer to work with a SPA2102 adapter. > I don't think it has something to do with FS, but I'm hoping someone here > could help me. > I do not get a new line in the phone (by pressing the R button), all DTMF > tones are sent as audio to the other connected phone. > > Anyone got it working? > > Thanks, > Jonas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/630a8c1b/attachment-0002.html From mgende at gendesign.com Wed Dec 9 07:16:03 2009 From: mgende at gendesign.com (Michael Gende) Date: Wed, 9 Dec 2009 09:16:03 -0600 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B1EB50D020000E1000002C7@mail.fribert.dk> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> Message-ID: Hey There, I came in not seeing any former posts of yours, so if this one is unhelpful, just delete. I did my FS install using PFsense as well. Its been working famously for a few months now. Very happy. I wrote what I discovered for FS on PFsense in this wiki: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I assume since you're using PFsense that your computer is functioning as a firewall AND a phone system (a dual homed host, in other words). That's what the wiki attempt above is aimed at. If you follow those instructions, you'll send and receive calls (as we were and are able to). It is working for us, at least. One problem in your case: I didn't really like using the PFsense Web interface when configuring FS (except for installing FS and setting some system parameters. Its great for PFsense, though). It helped me more to get in with ssh and vi and make FS work. Having done that successfully, you'll be more likely to effectively use the PFsense web interface for FS, as it's really just a "short cut" for someone that understands the FS file system, in my opinion. Good luck, Mike G. On Tue, Dec 8, 2009 at 1:20 PM, mailinglist wrote: > Hi All > > Ok, after reading a bit more I think I see what I've done wrong, but I > don't know how to fix it properly. > Looking in the Dialplan directory I see the following: > default (dir) > default.xml > features.xml > public (dir) > public.xml > > Under the default dir the webinterface has created the 001_musimi.dk.xml > file that I've created. > But as I understand it, it doesn't use it. > > How do I make it use it, I would very much like to keep the webinterface > editor, and not have to do it via ssh and vi all the time. > > >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i > meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: > Hi Mark > > Ok, thanks. > Yes I have a gateway placed in external called musimi.dk (or should it be > in public?), and I'll just create the empty XML's in lan to get rid of that > error. > > I'll remove the second part of the dialplan, my idea was that it was needed > for calls between sip phones hooked up to the freeswitch. > > Now the remaining problem: > When I call ext 1002 from ext 1001 I see this message and get an error, the > same goes for dialing 0XXXXXXXX to get an external number: > > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing > 1001->1002 in context default > 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel > sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] > 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 > [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. > Cause: NO_ROUTE_DESTINATION > 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup > sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 > (sofia/external/$1) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/external/$1 [CS_DESTROY] > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( > sofia/internal/1001 at 10.11.12.25) Ended > 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] > I don't see any mention of the statements in the Dialplan, so for me it > looks like it haven't registered the Dialplan? > > Best regards > Kenneth > > >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen > 659603.29094.qm at web56408.mail.re3.yahoo.com: > > Question ---------------------------------------------- > If I do a reloadxml it gives me this output on the console: > freeswitch at firewall.fribert.dk> > reloadxml > 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > Error including > /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No > such file or directory) > > I'm not sure if it's a genuine problem,as I can see it, it just complains > that I haven't created any sip_profiles in /lan, but is that necessary? > > Response: ---------------------------------------------- > This isn't really a problem. To get rid of the error simply put a blank xml > file into each folder as in the internal and external directories. Dump the > lan directory and lan profile as mentioned earlier. > > Question ---------------------------------------------- > > Extension Name musimi.dk > Enabled true > Order 001 > Description ... > > condition ^0(.\d+)$ > action bridge sofia/gateway/musimi.dk/$1 > > Response: ---------------------------------------------- > > This is correct as long as you have a gateway that is registered called > musimi.dk > > Question ---------------------------------------------- > Extension Name 10.11.12.25 > Enabled true > Order 002 > Description ... > > action bridge sofia/internal/$ > > Response: ---------------------------------------------- > > No idea what this is for its not needed as far as I can tell. > > > Now please summarize what you still need help on. > > > Mark J Crane > http://fusionpbx.com > pfSense FreeSWITCH package developer > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/d59b1ce5/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Dec 9 07:29:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 9 Dec 2009 15:29:37 -0000 Subject: [Freeswitch-users] no hang-up on B leg In-Reply-To: <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> References: <87f2f3b90912071111r4b7c63d2i8314677064981af2@mail.gmail.com> <3E3B241A-8BF0-4DCE-AAB3-DCFC4D4354B2@jerris.com> Message-ID: I would have tended to agree with the glare, however, before I killed both sides, I was back to my issue of the call not clearing down at all. (rtp timeout eventually does it) Thanks for the pointer to the source. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 09 December 2009 14:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hang-up on B leg src/switch_ivr_bridge.c This could just as well be a glare condition when the call is in process of tearing down. Mike On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote: No doubt, but that's a little difficult as this only happens occasionally and I have 200 calls going on at the time. It's needle in the haystack stuff. Here's what I know. I have an external process listening for DTMF events. If I detect '*' I do a kill uuid on the B leg. On a number of occasions I get an error saying the B leg doesn't exist, so I now do a double kill on the associated leg which I get from the event. I do not get a 'doesn't exist' message for the A leg, which leads me to believe that process of tearing down both bridged legs is flawed. The kluge clears the B leg hang issue, so the pressure's off for me, but when I get a few nano seconds, I'll look at the code to see if there's anything obvious. Can anyone give me a hint on what module handles bridged calls? (sorry, being lazy and suffering from a lack of sleep) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 08 December 2009 16:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/e391ec1a/attachment-0002.html From fernando.testa at gmail.com Wed Dec 9 07:25:11 2009 From: fernando.testa at gmail.com (Fernando Testa) Date: Wed, 9 Dec 2009 13:25:11 -0200 Subject: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones In-Reply-To: References: Message-ID: <22F7B75B-6568-4863-A7ED-165A800A928D@gmail.com> It worked! Tnx! Em 08/12/2009, ?s 16:51, Brian West escreveu: > Best option for you is to use 96 in the sofia profile you're using to > talk to these broken devices. > > /b > > On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: > >> Dear list, >> >> Some Nec phones sends DTMF RFC2833 with payload 101 during the call, >> but have negotiated a different one on SDP. >> When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 >> we notice this phone sends the following INVITE packet and RTP >> packets: http://pastebin.freeswitch.org/11433 >> Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap >> >> Is there any parameter to tweak FS in such a way to force understand >> 101 packets as DTMF? >> Thank you in advance! >> >> Fernando Testa >> PS: On pcap you have the following IPs: >> FS at 10.91.10.210 >> Nec Pbx 10.91.10.22 >> phone 10.91.10.85 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lgraybill at izeni.com Wed Dec 9 08:50:03 2009 From: lgraybill at izeni.com (Luke Graybill) Date: Wed, 9 Dec 2009 09:50:03 -0700 Subject: [Freeswitch-users] controlling calls handled within a fifo using event_socket Message-ID: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> In my FreeSWITCH environment, calls are originated out to customers who are placed into a fifo upon answer. There are members (x-lite endpoints) in this fifo who handle those customer calls. I am writing a monitoring application that uses event_socket to watch the channels involved in this process, ultimately displaying an interface for each rep that allows them to interactively drive the calls (playback audio conditionally to the customer, save information obtained during the call to another database, etc). Problems arise when attempting to identify which customer channel is speaking to which rep (consumer) channel. My event_socket application is inspecting the CHANNEL_ANSWER event, but this event does not appear to contain enough information to make this determination. I have identified three distinct uuid values in the CHANNEL_ANSWER headers on the consumer channel: core uuid, the uuid of the consumer channel, and another uuid which is not the customer uuid (I'm assuming this is the uuid of the fifo). According to the wiki here, I expected the consumer CHANNEL_ANSWER headers to contain variables such as `fifo_target` with the uuid of the customer channel it is bridged to, but this variable is not in the headers. Indeed, no channel variables are set which correspond to the uuid of the customer channel to which the rep is speaking. After the call has been completed, data posted in the cdr does in fact contain the `fifo_target` information, but this does not help me during the call. The short version of my question is this: how do I programmatically determine which channel uuid the consumer channel in a fifo is connected to? Any help here would be greatly appreciated :) Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/5aed75f9/attachment-0002.html From brian at freeswitch.org Wed Dec 9 08:55:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 10:55:36 -0600 Subject: [Freeswitch-users] controlling calls handled within a fifo using event_socket In-Reply-To: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> References: <83c6838b0912090850o77cb389fvad2ee35cd2fa1860@mail.gmail.com> Message-ID: <68510DE6-DBCF-406E-92E9-8C67B29AF59F@freeswitch.org> "fifo list" issue this API and get the fifo XML and get the caller's uuid out of the list. /b On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote: > The short version of my question is this: how do I programmatically > determine which channel uuid the consumer channel in a fifo is > connected to? From Prometheus001 at gmx.net Wed Dec 9 09:13:55 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 09 Dec 2009 18:13:55 +0100 Subject: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer Message-ID: <4B1FDAD3.8020305@gmx.net> Hello, in our dialplan we have enabled multiple-registrations, so 2 phones can register on a single directory entry. Both phones are registered, both phones can be called and each phone can call the other phone. However in an attended_transfer mode calls cannot be transferred to the other phone with the same number. Attended_transfer in this case is needed when you take a call on your main SIP phone and and then want to transfer it to your mobile DECT/SIP phone, because you may have to check something in another room. I did a SIP trace and see the following: * A invites B(phone 1) => ok * B(phone 1) places call on hold => ok * B(phone 1) dials number B(phone 2 DECT) on second line * Freeswitch send Invite to B(phone 1) => ok * Freeswitch send Invite to B(phone 2 DECT) * B(phone 2 DECT) sends Ringing to Freeswitch => ok * B(phone 1) sends Busy to Freeswitch * B(phone 1) displays Busy and hangs up the second line Is there any way to overcome this? Is there a way to ignore the Busy from phone 1 when phone 2 answers Ringing? Best regards Peter From edpimentl at gmail.com Wed Dec 9 09:29:11 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 9 Dec 2009 12:29:11 -0500 Subject: [Freeswitch-users] esl for Mac OS X 10.4 In-Reply-To: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> References: <5870116D-A2C1-46FE-BD56-94310E4430D9@jerris.com> Message-ID: <9dc4a1670912090929x1193c69du6cb47ec4da479b39@mail.gmail.com> Regarding Mac OSX 10.5/6 can you point me where the latest "FS binary" file is? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/62930daf/attachment-0002.html From msc at freeswitch.org Wed Dec 9 09:41:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Dec 2009 09:41:21 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: Message-ID: <87f2f3b90912090941p245d788cr4c530e17f88e162f@mail.gmail.com> On Tue, Dec 8, 2009 at 3:59 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Thought I?d send this little hurrah! As there seems to have been a lot > of negativity on this list lately. > > > I hereby multiply all the negative comments by -1. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/36dccb24/attachment-0002.html From jmesquita at freeswitch.org Wed Dec 9 10:37:37 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 9 Dec 2009 16:37:37 -0200 Subject: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer In-Reply-To: <4B1FDAD3.8020305@gmx.net> References: <4B1FDAD3.8020305@gmx.net> Message-ID: That is more dependent on the endpoint than on the switch itself. I guess you can always use mod_limit to come up with some crazy key to identify one endpoint or the other but still it seems overly complicated for something that is not supposed to be working this way. You can also park the call instead of transferring, can't ya? JM On Wed, Dec 9, 2009 at 3:13 PM, Peter P GMX wrote: > Hello, > > in our dialplan we have enabled multiple-registrations, so 2 phones can > register on a single directory entry. > > Both phones are registered, both phones can be called and each phone can > call the other phone. > However in an attended_transfer mode calls cannot be transferred to the > other phone with the same number. > Attended_transfer in this case is needed when you take a call on your > main SIP phone and and then want to transfer it to your mobile DECT/SIP > phone, because you may have to check something in another room. > I did a SIP trace and see the following: > > * A invites B(phone 1) => ok > * B(phone 1) places call on hold => ok > * B(phone 1) dials number B(phone 2 DECT) on second line > * Freeswitch send Invite to B(phone 1) => ok > * Freeswitch send Invite to B(phone 2 DECT) > * B(phone 2 DECT) sends Ringing to Freeswitch => ok > * B(phone 1) sends Busy to Freeswitch > * B(phone 1) displays Busy and hangs up the second line > > Is there any way to overcome this? Is there a way to ignore the Busy > from phone 1 when phone 2 answers Ringing? > > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/3c76a285/attachment-0002.html From codecomplete at free.fr Wed Dec 9 11:54:51 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 9 Dec 2009 11:54:51 -0800 (PST) Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: Message-ID: <26716612.post@talk.nabble.com> Nik Middleton wrote: > I cannot imagine doing what I'm using FS for, with any other product. Yes > it's frustrating at times, but this is largely down to a lack > documentation/samples. Speaking of which... would this layout be good for a book on Freeswitch? Preface 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. 2. Choosing hardware options (server, phones, gateways) 3. Setting up FS 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS gateways, etc.) 5. Administering FS (CLI and GUI) 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) 7. Performance, sound quality, other issues 8. Writing scripts (LUA, etc.), connecting to databases 9. Real-life examples (Gino's Pizza, etc.) Conclusion Index -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p26716612.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dftoro at yahoo.com Wed Dec 9 12:04:56 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 9 Dec 2009 12:04:56 -0800 (PST) Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: Message-ID: <771079.9902.qm@web33508.mail.mud.yahoo.com> I had hear about Welltech (http://www.welltech.com/default.aspx) gateways but I don't have any experience with them. Someone know ?, any experience... Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 11/25/09, Milena wrote: From: Milena Subject: Re: [Freeswitch-users] Grandstream gateways To: freeswitch-users at lists.freeswitch.org Date: Wednesday, November 25, 2009, 4:00 PM Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti wrote: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. ?I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews wrote: > Samuel - you could go with FXS gateways or channel banks. ?If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. ?If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. ?Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. ?Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? ?Please share your experience > with my boss, ?Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. ?I wanted assistance with choosing the most appropriate > hardware. ?We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 ?FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > ?wrote: > >> Send FreeSWITCH-users mailing list submissions to >> ? freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> ? freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> ? freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> ?1. Re: mod_conference kick to abort invitations (Michael Jerris) >> ?2. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Michael Jerris) >> ?3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> ?4. Re: remote_media_ip variable not set (Michael Jerris) >> ?5. Re: How to find whether the destination ? ?extension supports >> ? ? encryption (Michael Jerris) >> ?6. Re: Bypass_media and re_invite (srinivasula reddy) >> ?7. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Stephen Crosby) >> ?8. Re: Handling the 302 Moved Temporarily response ? ?from >> ? ? JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> ? invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an ?invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. ?This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> ? proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. ?I just added some code to set it on auto- >> adjust so it might be there sometimes now. ?You might need to add >> some code in mod_sofia to add it other times. ?Maybe it makes sense >> to move that var setting down to switch_rtp.c. ?Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris >> Subject: Re: [Freeswitch-users] How to find whether the destination >> ? extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the ?first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. ?i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. ?This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> ? response ? ?from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> ? <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> ? >> ? ? > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> ? ? ? ? >> ? ? ? ? >> >> ? ? ? ?> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> ? ? ? ?> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> ? ? ? ?> data="aPfx=${caller_id_number:0:6}"/> >> ? ? ? ?> data="aNum=${caller_id_number:6:16}"/> >> ? ? ? ?> data="IP_ADDR=${network_addr}:5060"/> >> >> ? ? ? ? >> >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> >> ? ? ? ? >> ? ? ? >> ? >> >> >> ? >> ? ? >> ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? >> >> ? >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. ?This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/1bf7a2f7/attachment-0002.html From ken at ksac.com Wed Dec 9 12:12:36 2009 From: ken at ksac.com (Kendall Stauffer) Date: Wed, 9 Dec 2009 12:12:36 -0800 Subject: [Freeswitch-users] Grandstream gateways In-Reply-To: <771079.9902.qm@web33508.mail.mud.yahoo.com> References: <771079.9902.qm@web33508.mail.mud.yahoo.com> Message-ID: Yes. I have one if anybody wants it, would let it go cheap. Works fine, but caller id is only the number, not the name part. Other than that works fine with astersik From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Toro Sent: Wednesday, December 09, 2009 3:05 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Grandstream gateways I had hear about Welltech (http://www.welltech.com/default.aspx) gateways but I don't have any experience with them. Someone know ?, any experience... Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 11/25/09, Milena wrote: From: Milena Subject: Re: [Freeswitch-users] Grandstream gateways To: freeswitch-users at lists.freeswitch.org Date: Wednesday, November 25, 2009, 4:00 PM Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media. Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen > You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control. This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users. On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti > wrote: Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews > wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews at sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> >> Today's Topics: >> >> 1. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii >> >> In trunk there is a sofia profile setting to allow dialplan >> processing of 302 responses. This won't get you back into your same >> javascript, but you can probably do something clever from there. >> >> Mike >> >> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >> >>> >>> I have considered writing JavaScript code to bridge two calls >>> together. However, I would like to perform custom handling of the >>> 302 Moved Temporarily response. How do I handle the 302 Moved >>> Temporarily response if I use JavaScript? >>> >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Nov 2009 11:46:05 -0600 >> From: Brian West > >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org> >> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes >> >> Yes an alias will be required for every domain you run on the profile >> so it can find it. >> >> /b >> >> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: >> >>> Try an alias on the sip profile. >>> >>> Mike >> >> >> >> >> ------------------------------ >> >> Message: 4 >> Date: Wed, 25 Nov 2009 12:47:37 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: > >> Content-Type: text/plain; charset=us-ascii >> >> It's possible it does not. I just added some code to set it on auto- >> adjust so it might be there sometimes now. You might need to add >> some code in mod_sofia to add it other times. Maybe it makes sense >> to move that var setting down to switch_rtp.c. Patches for this >> would be welcome. >> >> Thanks >> >> Mike >> >> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote: >> >>> Hi, >>> >>> In the case of proxy_media=true, does it gets set at all then? >> >> >> >> >> ------------------------------ >> >> Message: 5 >> Date: Wed, 25 Nov 2009 12:48:39 -0500 >> From: Michael Jerris > >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com> >> Content-Type: text/plain; charset=us-ascii >> >> You can send the call with secure enabled and if it supports it it >> will use it. >> >> Mike >> >> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote: >> >>> Hello, >>> >>> We have a mix of phones that support RTP encryption and those that >>> do not. I have to support both types in the meanwhile, and would >>> like to have encryption enabled on the relevant leg, even if the >>> other leg does not support it (why? one of our ATAs either must >>> have it unencrypted or have it encrypted, but cannot have both). >>> >>> How do I find whether the destination supports encryption? I do not >>> want to manage an additional table in the database... >>> >> >> >> >> ------------------------------ >> >> Message: 6 >> Date: Wed, 25 Nov 2009 23:25:01 +0530 >> From: srinivasula reddy > >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> > >> Content-Type: text/plain; charset="iso-8859-1" >> >> HI, >> thanks for your reply, my requirement is i am doing failover stuff >> with >> freeswitch. i dont want cut the calls when freeswitch dies, when >> failover >> happens mean one freeswitch dies we are going to start the second >> freeswitch, i dont want close call intiated by the first >> freeswtich, they >> are communicating with meida(bypass media). when one endpoing try to >> end the >> call at that time i want to close the call for the other end also. >> >> >> srinivas >> >> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris > >> wrote: >> >>> FreeSWITCH will kill the calls when you shut it down, if you >>> intentionally >>> kill the network without shutting down FreeSWITCH the only thing >>> you can do >>> is enable session timers or rtp timers in the soft phones to kill >>> the call >>> when FreeSWITCH dies or when the call is over. >>> >>> Mike >>> >>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote: >>> >>>> Hi All, >>>> >>>> goodmorning to all, i have a scenario, two pjsua clients are >>>> connected >>> with Freeswitch and they are in call and bypass_media=true. i >>> close the >>> Freeswitch server, still they are in call, again i started the >>> Freeswitch, >>> and registerd these two endpoints, now how can i end the call >>> (estabilished >>> by the first Freeswitch)? if i call re_invite will it estabilish >>> the call >>> between two endpoints? >>>> any idea? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Srinivasula Reddy K >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Surprisingly, I've found no way to access the HTTP response status >> code >> using mod_spidermonkey_curl. I'd love to see this feature added or >> discussed >> if it already exists and I'm missing it. >> >> --Stephen >> >> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris > >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga > >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com> >> Content-Type: text/plain; charset="iso-8859-1" >> >> this is how i do it from the dialplan: >> >> >> >> >> >> > expression="^(300030)(.*)|^\+(300030)(.*)"> >> >> >> >> >> > data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/> >> > data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: >> 1:32} : >> ${caller_id_number})}"/> >> >> > data="aPfx=${caller_id_number:0:6}"/> >> > data="aNum=${caller_id_number:6:16}"/> >> > data="IP_ADDR=${network_addr}:5060"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris > >> wrote: >> >>> In trunk there is a sofia profile setting to allow dialplan >>> processing of >>> 302 responses. This won't get you back into your same javascript, >>> but you >>> can probably do something clever from there. >>> >>> Mike >>> >>> On Nov 24, 2009, at 5:04 PM, John Platts wrote: >>> >>>> >>>> I have considered writing JavaScript code to bridge two calls >>>> together. >>> However, I would like to perform custom handling of the 302 Moved >>> Temporarily response. How do I handle the 302 Moved Temporarily >>> response if >>> I use JavaScript? >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189 >> ************************************************* > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/751d0ab1/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Dec 9 12:50:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 9 Dec 2009 20:50:01 -0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26716612.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> Message-ID: Looks good, but you've missed out billing and the key one, the event socket which could be a chapter in it's self. Do you have a publisher for it yet? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 Sent: 09 December 2009 19:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Rocks!!!!!!!!! Nik Middleton wrote: > I cannot imagine doing what I'm using FS for, with any other product. Yes > it's frustrating at times, but this is largely down to a lack > documentation/samples. Speaking of which... would this layout be good for a book on Freeswitch? Preface 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. 2. Choosing hardware options (server, phones, gateways) 3. Setting up FS 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS gateways, etc.) 5. Administering FS (CLI and GUI) 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) 7. Performance, sound quality, other issues 8. Writing scripts (LUA, etc.), connecting to databases 9. Real-life examples (Gino's Pizza, etc.) Conclusion Index -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p267 16612.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mailinglist at fribert.dk Wed Dec 9 13:46:00 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:46:00 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through Message-ID: <4B2028A8020000E10000030E@mail.fribert.dk> Hi Michael Thankyou for the excellent wiki article, yes, I did follow your guide there, all the way except to 'dialplan' and it seems that's the problem at the moment. I would very much like to create the dialplan in the webinterface, and not in the public.xml file. But at the moment it only uses the public.xml file :-( Great writeup you made, and it has brought me a long way. BR Fribert >>> 09-12-2009 kl. 16:16 skrev Michael Gende i meddelelsen : Hey There, I came in not seeing any former posts of yours, so if this one is unhelpful, just delete. I did my FS install using PFsense as well. Its been working famously for a few months now. Very happy. I wrote what I discovered for FS on PFsense in this wiki: http://wiki.freeswitch.org/wiki/Multi_home_tutorial I assume since you're using PFsense that your computer is functioning as a firewall AND a phone system (a dual homed host, in other words). That's what the wiki attempt above is aimed at. If you follow those instructions, you'll send and receive calls (as we were and are able to). It is working for us, at least. One problem in your case: I didn't really like using the PFsense Web interface when configuring FS (except for installing FS and setting some system parameters. Its great for PFsense, though). It helped me more to get in with ssh and vi and make FS work. Having done that successfully, you'll be more likely to effectively use the PFsense web interface for FS, as it's really just a "short cut" for someone that understands the FS file system, in my opinion. Good luck, Mike G. On Tue, Dec 8, 2009 at 1:20 PM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/32371c67/attachment-0002.html From mailinglist at fribert.dk Wed Dec 9 13:47:04 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:47:04 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <187489.95329.qm@web56408.mail.re3.yahoo.com> References: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> <187489.95329.qm@web56408.mail.re3.yahoo.com> Message-ID: <4B2028E8020000E100000313@mail.fribert.dk> This is a new install, but it's grabbed from a pfSense repository. >>> 09-12-2009 kl. 10:28 skrev Mark Crane i meddelelsen <187489.95329.qm at web56408.mail.re3.yahoo.com>: Is this a new install of the FreeSWITCH package or is it an upgrade from and earlier package? Mark J Crane mctch at yahoo.com --- On Tue, 12/8/09, Nandy Dagondon wrote: From: Nandy Dagondon Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Tuesday, December 8, 2009, 3:45 PM have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk ( /mc/compose?to=4B1DFABC020000E1000002C2 at mail.fribert.dk )>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 )) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 ( /mc/compose?to=sofia/internal/1001 at 10.11.12.25 ) [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com ( /mc/compose?to=659603.29094.qm at web56408.mail.re3.yahoo.com ): Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/bedc27ab/attachment-0002.html From mailinglist at fribert.dk Wed Dec 9 13:47:52 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 22:47:52 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <4B202918020000E100000318@mail.fribert.dk> Yes, I have two extensions. I can even make them join a group, and if I call the group, the two extensions will ring. >>> 08-12-2009 kl. 23:45 skrev Nandy Dagondon i meddelelsen <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4 at mail.gmail.com>: have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. >>> 08-12-2009 kl. 07:05 skrev "mailinglist" i meddelelsen <4B1DFABC020000E1000002C2 at mail.fribert.dk>: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0XXXXXXXX to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001->1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1001 at 10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1001 at 10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth >>> 08-12-2009 kl. 03:05 skrev Mark Crane i meddelelsen 659603.29094.qm at web56408.mail.re3.yahoo.com: Question ---------------------------------------------- If I do a reloadxml it gives me this output on the console: freeswitch at firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswitch at firewall.fribert.dk )> reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: ---------------------------------------------- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question ---------------------------------------------- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: ---------------------------------------------- This is correct as long as you have a gateway that is registered called musimi.dk Question ---------------------------------------------- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: ---------------------------------------------- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/58f3afc3/attachment-0002.html From timuckun at gmail.com Wed Dec 9 13:56:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 10:56:31 +1300 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26716612.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> Message-ID: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> > > Preface > 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc. > 2. Choosing hardware options (server, phones, gateways) > 3. Setting up FS > 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS > gateways, etc.) > 5. Administering FS (CLI and GUI) > 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.) > 7. Performance, sound quality, other issues > 8. Writing scripts (LUA, etc.), connecting to databases > 9. Real-life examples (Gino's Pizza, etc.) > Conclusion > Index > -- I found the rosetta stone useful though woefully lacking in volume. I guess that's true overall with the project. From brian at freeswitch.org Wed Dec 9 14:07:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 16:07:28 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> Message-ID: <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> Visit the friday meetings and we can help if you document it. ;) /b On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote: > I found the rosetta stone useful though woefully lacking in volume. > > I guess that's true overall with the project. From msc at freeswitch.org Wed Dec 9 14:10:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Dec 2009 14:10:35 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> Message-ID: <87f2f3b90912091410p322a267cne481079856f74ed5@mail.gmail.com> > I found the rosetta stone useful though woefully lacking in volume. > > I guess that's true overall with the project. > > Documentation is neither easy nor glamorous. The woefully lacking documentation has been provided by a little group of people who've done a big bit of documenting and a big group of people who've done a little bit of documenting. If ever there was an aspect of this project that could use more volunteers it is documentation and bug testing. If anyone wants to help on either of these fronts please email me off list. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/5906c1be/attachment-0002.html From mailinglist at fribert.dk Wed Dec 9 14:20:01 2009 From: mailinglist at fribert.dk (mailinglist) Date: Wed, 09 Dec 2009 23:20:01 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through References: <4B1D78AF020000E1000002BC@mail.fribert.dk> <659603.29094.qm@web56408.mail.re3.yahoo.com> <4B1DFABC020000E1000002C2@mail.fribert.dk> <4B1EB50D020000E1000002C7@mail.fribert.dk> <7d0bfd8c0912081445v124dd6cs9174a201eb1096a4@mail.gmail.com> Message-ID: <4B2030A1020000E10000031D@mail.fribert.dk> WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/6f3edd80/attachment-0002.html From timuckun at gmail.com Wed Dec 9 14:35:31 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 11:35:31 +1300 Subject: [Freeswitch-users] Even socket question. Message-ID: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> Hey All. I am trying to get freeswitch to route to my socket handler and am having a problem. I am running freeswitch inside a virtualbox VM for testing purposes. The vitualbox communicates with my host via the "host only" adapter. The VM IP address is 192.168.56.3 and the laptop has the iP 192.168.56.1 I have set up both an outbound and an inbound socket handlers. The inbound one works fine, the outbound is not working . The inbound merely logs the event name. The outbound logs the connection and hangs up. I have set up an extension like this When I dial 8084 I get a lot of events being logged but the oubound never gets the calls and never logs the call. I have added the fs_cli output below. It looks to me like it's sending the output to the other IP address of my laptop instead of the one I specified in my extension but I could just be misreading that. I have set the external IP of the freeswitch to the 56.3 address. Here is the LSOF output freeswitc 2468 root 31u IPv4 5785 TCP ubuntuvm01:5080 (LISTEN) freeswitc 2468 root 33u IPv6 5791 TCP localhost:5060 (LISTEN) freeswitc 2468 root 36u IPv4 5804 TCP 192.168.56.3:5060 (LISTEN) freeswitc 2468 root 48u IPv4 5910 TCP 192.168.56.3:8021 (LISTEN) freeswitc 2468 root 50u IPv4 5912 TCP *:8080 (LISTEN) Here is the output from the fs_cli 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl "domains". Falling back to Digest auth. 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0] 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1 Rejected by acl "domains". Falling back to Digest auth. 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1000 at 192.168.56.3 [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa] 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_NEW 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000 at 192.168.56.3) State NEW 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [received][100] 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP: v=0 o=Z 0 0 IN IP4 218.101.6.157 s=Z c=IN IP4 218.101.6.157 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:115:32000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G7221:107:16000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[G722:9:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMU:0:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[PCMA:8:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20] 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec sofia/internal/1000 at 192.168.56.3 GSM/8000 20 ms 160 samples 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf payload to 101 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885 (sofia/internal/1000 at 192.168.56.3) State Change CS_NEW -> CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_INIT 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.56.3) State INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 192.168.56.3 SOFIA INIT 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111 (sofia/internal/1000 at 192.168.56.3) State Change CS_INIT -> CS_ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000 at 192.168.56.3) State INIT going to sleep 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.56.3) State ROUTING 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:132 sofia/internal/1000 at 192.168.56.3 SOFIA ROUTING 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1000 at 192.168.56.3 Standard ROUTING 2009-12-09 14:31:53.423898 [INFO] mod_dialplan_xml.c:408 Processing 1000->8084 in context default Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->unloop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->tod_example] continue=true Dialplan: day of week[4] =~ 2-6 (PASS) Dialplan: hour[14] =~ 9-18 (PASS) Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Action set(open=true) Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->holiday_example] continue=true Dialplan: month[12] =~ 1 (FAIL) Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global-intercept] destination_number(8084) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [group-intercept] destination_number(8084) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [intercept-ext] destination_number(8084) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->redial] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [redial] destination_number(8084) =~ /^870$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global] continue=true Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000 at 192.168.56.3 Absolute Condition [global] Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-2] destination_number(8084) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-1] destination_number(8084) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] destination_number(8084) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] destination_number(8084) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->call_return] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [call_return] destination_number(8084) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->del-group] continue=false Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [del-group] destination_number(8084) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000 at 192.168.56.3 Action answer() Dialplan: sofia/internal/1000 at 192.168.56.3 Action group(delete:84@${domain_name}:${sofia_contact(${sip_from_user}@${domain_name})}) Dialplan: sofia/internal/1000 at 192.168.56.3 Action gentones(%(1000, 0, 320)) 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:122 (sofia/internal/1000 at 192.168.56.3) State Change CS_ROUTING -> CS_EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000 at 192.168.56.3) State ROUTING going to sleep 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.56.3) State EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:181 sofia/internal/1000 at 192.168.56.3 SOFIA EXECUTE 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:159 sofia/internal/1000 at 192.168.56.3 Standard EXECUTE EXECUTE sofia/internal/1000 at 192.168.56.3 set(open=true) 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:768 sofia/internal/1000 at 192.168.56.3 SET [open]=[true] EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-spymap/1000/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-last_dial/1000/8084) EXECUTE sofia/internal/1000 at 192.168.56.3 hash(insert/192.168.56.3-last_dial/global/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) EXECUTE sofia/internal/1000 at 192.168.56.3 answer() 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:658 sofia/internal/1000 at 192.168.56.3 receive message [ANSWER] 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:2381 AUDIO RTP [sofia/internal/1000 at 192.168.56.3] 192.168.50.173 port 27042 -> 218.101.6.157 port 8000 codec: 3 ms: 20 2009-12-09 14:31:53.423898 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 160 bytes per 20ms 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:571 Local SDP sofia/internal/1000 at 192.168.56.3: v=0 o=FreeSWITCH 1260370871 1260370872 IN IP4 192.168.50.173 s=FreeSWITCH c=IN IP4 192.168.50.173 t=0 0 m=audio 27042 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:645 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:53.423898 [NOTICE] mod_dptools.c:658 Channel [sofia/internal/1000 at 192.168.56.3] has been answered 2009-12-09 14:31:53.423898 [DEBUG] switch_channel.c:182 sofia/internal/1000 at 192.168.56.3 receive message [AUDIO_SYNC] EXECUTE sofia/internal/1000 at 192.168.56.3 group(delete:84 at 192.168.56.3:sofia/internal/sip:1000 at 218.101.6.157:5070;rinstance=a8b6fdbc731e3b66;transport=UDP) EXECUTE sofia/internal/1000 at 192.168.56.3 gentones(%(1000, 0, 320)) 2009-12-09 14:31:53.436374 [DEBUG] switch_core_io.c:652 sofia/internal/1000 at 192.168.56.3 receive message [TRANSCODING_NECESSARY] 2009-12-09 14:31:53.436670 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [completed][200] 2009-12-09 14:31:53.490803 [DEBUG] sofia.c:3727 Channel sofia/internal/1000 at 192.168.56.3 entering state [ready][200] 2009-12-09 14:31:53.729534 [INFO] switch_rtp.c:1987 Auto Changing port from 218.101.6.157:8000 to 192.168.50.105:8000 2009-12-09 14:31:54.430526 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/1000 at 192.168.56.3 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-09 14:31:54.430526 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/1000 at 192.168.56.3 [KILL] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 (sofia/internal/1000 at 192.168.56.3) State HANGUP 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:358 Channel sofia/internal/1000 at 192.168.56.3 hanging up, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:400 Sending BYE to sofia/internal/1000 at 192.168.56.3 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000 at 192.168.56.3 Standard HANGUP, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 (sofia/internal/1000 at 192.168.56.3) State HANGUP going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000 at 192.168.56.3) State EXECUTE going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_HANGUP 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:465 sofia/internal/1000 at 192.168.56.3 handler already called, skipping state handler. 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000 at 192.168.56.3) State Change CS_HANGUP -> CS_REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/1000 at 192.168.56.3) State REPORTING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000 at 192.168.56.3 Standard REPORTING, cause: NORMAL_CLEARING 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 (sofia/internal/1000 at 192.168.56.3) State REPORTING going to sleep 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/1000 at 192.168.56.3) State Change CS_REPORTING -> CS_DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send signal sofia/internal/1000 at 192.168.56.3 [BREAK] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:1136 Session 6 (sofia/internal/1000 at 192.168.56.3) Locked, Waiting on external entities 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/internal/1000 at 192.168.56.3) Ended 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1156 Close Channel sofia/internal/1000 at 192.168.56.3 [CS_DESTROY] 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1000 at 192.168.56.3) Running State Change CS_DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 192.168.56.3) State DESTROY 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:293 sofia/internal/1000 at 192.168.56.3 SOFIA DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000 at 192.168.56.3 Standard DESTROY 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 192.168.56.3) State DESTROY going to sleep From timuckun at gmail.com Wed Dec 9 14:39:26 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 11:39:26 +1300 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> Message-ID: <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> On Thu, Dec 10, 2009 at 11:07 AM, Brian West wrote: > Visit the friday meetings and we can help if you document it. ?;) > I would be willing to lend a hand with the documentation but I know so little (a complete freeswitch noob). For example I was trying to figure out how to tell if an extension was set up "show dialplan in asterisk". I could not find this anywhere. If I find out I would be happy to add it to the rosetta stone. I am currently working on getting outbound socket working. Once I get it going I would be happy to add it to the relevant section of the wiki (in this case ruby). From brian at freeswitch.org Wed Dec 9 14:43:34 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 16:43:34 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> Message-ID: <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> That is what is nice about our community I'm more than willing to answer the questions if you document them... as are many others in the core team...we just have a lot to do and I think the best repayment is documentation! ;) /b On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > On Thu, Dec 10, 2009 at 11:07 AM, Brian West > wrote: >> Visit the friday meetings and we can help if you document it. ;) >> > > I would be willing to lend a hand with the documentation but I know so > little (a complete freeswitch noob). For example I was trying to > figure out how to tell if an extension was set up "show dialplan in > asterisk". I could not find this anywhere. If I find out I would be > happy to add it to the rosetta stone. > > I am currently working on getting outbound socket working. Once I get > it going I would be happy to add it to the relevant section of the > wiki (in this case ruby). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/f0155fb5/attachment-0002.html From mctch at yahoo.com Wed Dec 9 15:00:36 2009 From: mctch at yahoo.com (Mark Crane) Date: Wed, 9 Dec 2009 15:00:36 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B2030A1020000E10000031D@mail.fribert.dk> Message-ID: <110796.60596.qm@web56401.mail.re3.yahoo.com> Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. ????? Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! ? It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? ? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml ? ? The default.xml looks like this ( I haven't changed it): ? ? ? ??? ????? ????? ? ????? ??? ? ??? ??? ????? ????? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ????? ? ????? ????? ????? ????? ? ? ? ????? ? ????? ? ? ? ????? ??? ? ??? ??? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ??? ??? ?? ??? ??? ????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ? ? ??? ??? ????? ? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ??? ????? ? ????? ??? ? ??? ?????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ??????? ??????? ? ????? ??? ? ??? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ????? ? ????? ??? ??? ??? ????? ????? ????? ? ? ????? ??? ? ??? ??? ????? ????? ????? ?]]> ? ????? ??? ??? ??? ??? ????? ????? ????? ? ? ?????? ??? ? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ??????? ??????? ????? ??? ? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ? ??? ????? ????? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ??? ??? ? ? ??? ??? ????? ??? ? ? ??? ??? ??? ? ??? ? ? ? ? Then I have under default dir: musimidk.xml ?? ?????? ?? and 9000_recordings.xml ?? ?????? ?? ? ? Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/6e5a4446/attachment-0002.html From brian at freeswitch.org Wed Dec 9 15:03:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Dec 2009 17:03:02 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Bug Reports... Message-ID: Dear FreeSWITCHers, As of Friday Dec. 11th we will NOT accept any more bug reports on 1.0.4. You need to be on a 1.0.5pre or SVN trunk. 1.0.4 is over 6 months old and I really suspect your issues in 1.0.4 are already fixed. We will release a new pre every monday morning till 1.0.5 is released please keep up to date if possible. We are working hard to get 1.0.5 out and be as stable as possible and its more stable than 1.0.4... their might be some edge or corner cases that aren't accounted for so we need you to please download SVN trunk in your test labs and try it out... report issues and help us make the best FreeSWITCH release possible. Thank you, Brian West From djbinter at yahoo.com Wed Dec 9 15:07:32 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 9 Dec 2009 15:07:32 -0800 (PST) Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. In-Reply-To: <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> References: <168319.49226.qm@web37502.mail.mud.yahoo.com> <987536.45831.qm@web37508.mail.mud.yahoo.com> <855e4dcf0912090226u529d2ffs6059b01c38d11ee8@mail.gmail.com> Message-ID: <769416.34727.qm@web37508.mail.mud.yahoo.com> Load sharing feature is coming off our Lucent Telica switch. ________________________________ From: Tim Uckun To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 9, 2009 2:26:41 AM Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. On Tue, Dec 8, 2009 at 5:42 AM, DJB wrote: > One thing that I forgot to mention, these 2 FreeSWITCH servers are getting > calls with load balancing from another switch. Thus, the traffic type are > pretty much identical and both FSs have exactly the same on configuration. > Any suggestion would be appreciated. Thank you. If you could explain how you are doing the load balancing it would be really helpful to me. I am trying to do the same thing. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/75713a81/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 9 17:46:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Dec 2009 19:46:07 -0600 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> Message-ID: <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> do you have something listening on 8084 ? On Wed, Dec 9, 2009 at 4:35 PM, Tim Uckun wrote: > Hey All. I am trying to get freeswitch to route to my socket handler > and am having a problem. > > I am running freeswitch inside a virtualbox VM for testing purposes. > The vitualbox communicates with my host via the "host only" adapter. > The VM IP address is 192.168.56.3 and the laptop has the iP > 192.168.56.1 > > I have set up both an outbound and an inbound socket handlers. The > inbound one works fine, the outbound is not working . The inbound > merely logs the event name. The outbound logs the connection and hangs > up. > > I have set up an extension like this > > > > > > > > > > > > > When I dial 8084 I get a lot of events being logged but the oubound > never gets the calls and never logs the call. > > I have added the fs_cli output below. It looks to me like it's sending > the output to the other IP address of my laptop instead of the one I > specified in my extension but I could just be misreading that. I > have set the external IP of the freeswitch to the 56.3 address. > > Here is the LSOF output > > freeswitc 2468 root 31u IPv4 5785 > TCP ubuntuvm01:5080 (LISTEN) > freeswitc 2468 root 33u IPv6 5791 > TCP localhost:5060 (LISTEN) > freeswitc 2468 root 36u IPv4 5804 > TCP 192.168.56.3:5060 (LISTEN) > freeswitc 2468 root 48u IPv4 5910 > TCP 192.168.56.3:8021 (LISTEN) > freeswitc 2468 root 50u IPv4 5912 > TCP *:8080 (LISTEN) > > > Here is the output from the fs_cli > > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0] > 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0] > 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/1000 at 192.168.56.3 > [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa] > 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_NEW > 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320 > (sofia/internal/1000 at 192.168.56.3) State NEW > 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [received][100] > 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP: > v=0 > o=Z 0 0 IN IP4 218.101.6.157 > s=Z > c=IN IP4 218.101.6.157 > t=0 0 > m=audio 8000 RTP/AVP 3 110 98 8 0 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:110 speex/8000 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=30 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G7221:115:32000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G7221:107:16000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[G722:9:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[PCMU:0:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[PCMA:8:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec > Compare [GSM:3:8000:20]/[GSM:3:8000:20] > 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec > sofia/internal/1000 at 192.168.56.3 GSM/8000 20 ms 160 samples > 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf > payload to 101 > 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885 > (sofia/internal/1000 at 192.168.56.3) State Change CS_NEW -> CS_INIT > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_INIT > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.56.3) State INIT > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83 > sofia/internal/1000 at 192.168.56.3 SOFIA INIT > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111 > (sofia/internal/1000 at 192.168.56.3) State Change CS_INIT -> CS_ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/1000 at 192.168.56.3) State INIT going to sleep > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.56.3) State ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:132 > sofia/internal/1000 at 192.168.56.3 SOFIA ROUTING > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/1000 at 192.168.56.3 Standard ROUTING > 2009-12-09 14:31:53.423898 [INFO] mod_dialplan_xml.c:408 Processing > 1000->8084 in context default > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Action set(open=true) > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/1000 at 192.168.56.3 Date/Time Match (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->global-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [global-intercept] destination_number(8084) =~ /^886$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->group-intercept] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [group-intercept] destination_number(8084) =~ /^\*8$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->intercept-ext] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) > [intercept-ext] destination_number(8084) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->redial] > continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [redial] > destination_number(8084) =~ /^870$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing [default->global] > continue=true > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > Dialplan: sofia/internal/1000 at 192.168.56.3 Absolute Condition [global] > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->snom-demo-2] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-2] > destination_number(8084) =~ /^9001$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->snom-demo-1] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [snom-demo-1] > destination_number(8084) =~ /^9000$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] > destination_number(8084) =~ /^88(.*)$|^\*0(.*)$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->eavesdrop] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [eavesdrop] > destination_number(8084) =~ /^779$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->call_return] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (FAIL) [call_return] > destination_number(8084) =~ /^\*69$|^869$|^lcr$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 parsing > [default->del-group] continue=false > Dialplan: sofia/internal/1000 at 192.168.56.3 Regex (PASS) [del-group] > destination_number(8084) =~ /^80(\d{2})$/ break=on-false > Dialplan: sofia/internal/1000 at 192.168.56.3 Action answer() > Dialplan: sofia/internal/1000 at 192.168.56.3 Action > group(delete:84@${domain_name}:${sofia_contact(${sip_from_user}@ > ${domain_name})}) > Dialplan: sofia/internal/1000 at 192.168.56.3 Action gentones(%(1000, 0, > 320)) > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/1000 at 192.168.56.3) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/1000 at 192.168.56.3) State ROUTING going to sleep > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.56.3) State EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:181 > sofia/internal/1000 at 192.168.56.3 SOFIA EXECUTE > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/1000 at 192.168.56.3 Standard EXECUTE > EXECUTE sofia/internal/1000 at 192.168.56.3 set(open=true) > 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:768 > sofia/internal/1000 at 192.168.56.3 SET [open]=[true] > EXECUTE sofia/internal/1000 at 192.168.56.3 > hash(insert/192.168.56.3-spymap/1000/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) > EXECUTE sofia/internal/1000 at 192.168.56.3 > hash(insert/192.168.56.3-last_dial/1000/8084) > EXECUTE sofia/internal/1000 at 192.168.56.3 > > hash(insert/192.168.56.3-last_dial/global/2fbcf6fe-b35e-4c40-92a6-9f21de3102fa) > EXECUTE sofia/internal/1000 at 192.168.56.3 answer() > 2009-12-09 14:31:53.423898 [DEBUG] mod_dptools.c:658 > sofia/internal/1000 at 192.168.56.3 receive message [ANSWER] > 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:2381 AUDIO RTP > [sofia/internal/1000 at 192.168.56.3] 192.168.50.173 port 27042 -> > 218.101.6.157 port 8000 codec: 3 ms: 20 > 2009-12-09 14:31:53.423898 [DEBUG] switch_rtp.c:1167 Starting timer > [soft] 160 bytes per 20ms > 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:571 Local SDP > sofia/internal/1000 at 192.168.56.3: > v=0 > o=FreeSWITCH 1260370871 1260370872 IN IP4 192.168.50.173 > s=FreeSWITCH > c=IN IP4 192.168.50.173 > t=0 0 > m=audio 27042 RTP/AVP 3 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:645 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:53.423898 [NOTICE] mod_dptools.c:658 Channel > [sofia/internal/1000 at 192.168.56.3] has been answered > 2009-12-09 14:31:53.423898 [DEBUG] switch_channel.c:182 > sofia/internal/1000 at 192.168.56.3 receive message [AUDIO_SYNC] > EXECUTE sofia/internal/1000 at 192.168.56.3 > group(delete:84 at 192.168.56.3:sofia/internal/sip:1000 at 218.101.6.157:5070 > ;rinstance=a8b6fdbc731e3b66;transport=UDP) > EXECUTE sofia/internal/1000 at 192.168.56.3 gentones(%(1000, 0, 320)) > 2009-12-09 14:31:53.436374 [DEBUG] switch_core_io.c:652 > sofia/internal/1000 at 192.168.56.3 receive message > [TRANSCODING_NECESSARY] > 2009-12-09 14:31:53.436670 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [completed][200] > 2009-12-09 14:31:53.490803 [DEBUG] sofia.c:3727 Channel > sofia/internal/1000 at 192.168.56.3 entering state [ready][200] > 2009-12-09 14:31:53.729534 [INFO] switch_rtp.c:1987 Auto Changing port > from 218.101.6.157:8000 to 192.168.50.105:8000 > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_state_machine.c:187 > Hangup sofia/internal/1000 at 192.168.56.3 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-12-09 14:31:54.430526 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/1000 at 192.168.56.3 [KILL] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 > (sofia/internal/1000 at 192.168.56.3) State HANGUP > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/1000 at 192.168.56.3 hanging up, cause: NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:400 Sending BYE to > sofia/internal/1000 at 192.168.56.3 > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/1000 at 192.168.56.3 Standard HANGUP, cause: > NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:488 > (sofia/internal/1000 at 192.168.56.3) State HANGUP going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/1000 at 192.168.56.3) State EXECUTE going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_HANGUP > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:465 > sofia/internal/1000 at 192.168.56.3 handler already called, skipping > state handler. > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/1000 at 192.168.56.3) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 > (sofia/internal/1000 at 192.168.56.3) State REPORTING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/1000 at 192.168.56.3 Standard REPORTING, cause: > NORMAL_CLEARING > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:579 > (sofia/internal/1000 at 192.168.56.3) State REPORTING going to sleep > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/1000 at 192.168.56.3) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:999 Send > signal sofia/internal/1000 at 192.168.56.3 [BREAK] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_session.c:1136 Session > 6 (sofia/internal/1000 at 192.168.56.3) Locked, Waiting on external > entities > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1154 Session > 6 (sofia/internal/1000 at 192.168.56.3) Ended > 2009-12-09 14:31:54.430526 [NOTICE] switch_core_session.c:1156 Close > Channel sofia/internal/1000 at 192.168.56.3 [CS_DESTROY] > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/1000 at 192.168.56.3) Running State Change CS_DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/1000 at 192.168.56.3) State DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] mod_sofia.c:293 > sofia/internal/1000 at 192.168.56.3 SOFIA DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/1000 at 192.168.56.3 Standard DESTROY > 2009-12-09 14:31:54.430526 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/1000 at 192.168.56.3) State DESTROY going to sleep > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/1c3e9f15/attachment-0002.html From timuckun at gmail.com Wed Dec 9 18:00:23 2009 From: timuckun at gmail.com (Tim Uckun) Date: Thu, 10 Dec 2009 15:00:23 +1300 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> Message-ID: <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale wrote: > do you have something listening on 8084 ? > Yes. I figured out the problem. There was already an extension called 8084 and it overwrote the extension I defined. Which brings me back to a question I had earlier. Where is the equivalent of the "show dialplan" command? How can I list all the extensions and their definitions? From anthony.minessale at gmail.com Wed Dec 9 18:12:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Dec 2009 20:12:07 -0600 Subject: [Freeswitch-users] Even socket question. In-Reply-To: <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> References: <855e4dcf0912091435s616f3f3dvfe7e9c89b8bf44a6@mail.gmail.com> <191c3a030912091746j5a5c95e9qe9b60fa0d6fd365d@mail.gmail.com> <855e4dcf0912091800t636d8430l527a8f7eb08d75e7@mail.gmail.com> Message-ID: <191c3a030912091812o5122b5cayd7e82a37ac224a43@mail.gmail.com> the dialplan is dynamic there is no such thing you have to look in your dialplan xml files because it's served up live. FS has a different paradigm than asterisk. On Wed, Dec 9, 2009 at 8:00 PM, Tim Uckun wrote: > On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale > wrote: > > do you have something listening on 8084 ? > > > > Yes. > > I figured out the problem. There was already an extension called 8084 > and it overwrote the extension I defined. > > Which brings me back to a question I had earlier. > > Where is the equivalent of the "show dialplan" command? How can I list > all the extensions and their definitions? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/99e3aa86/attachment-0002.html From brian at microcomaustralia.com.au Wed Dec 9 16:55:45 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 11:55:45 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware Message-ID: <20091210005545.GD28041@sys11.in.vpac.org> Hello, I asked this question on my local linux user group mailing list, and got the recommendation to ask here. Anyway, at the moment I am running Asterisk on an IP04 embedded system. http://www.rowetel.com/ucasterisk/ip04.html It works well most of the time, however there are some bugs that do, under circumstances lead to less then desirable behaviour (such as on some occasions which I don't fully understand sometimes the remote system fails to generate any audio packets when there is no audio - almost like silence suppression was supported by the remote system - and asterisk fails to generate any audio packets in return; on another slower computer running the same SIP software and on the same network everything works fine; as far as I can tell the software - twinkle - doesn't even support silence suppression). I suspect at least some - if not all - of the issues I have encountered may be resolved with Freeswitch, however I don't really want to replace my small, energy efficient, embedded system, with a large, power hungry computer system. Overkill. An added complication is I need at least 1 analogue port to connect to the Australian based telephone line (2 ports exchange ports and 1 extension port would be ideal but not essiential). Unfortunately, I have been told that the IP04 hardware isn't compatable with the requirements of Freeswitch. Such as not having a MMU. So there doesn't appear to be much effort porting Freeswitch to IP04 as a result. I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Are there any other good alternatives? Thanks. -- Brian May From kristian.kielhofner at gmail.com Wed Dec 9 19:47:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 9 Dec 2009 22:47:05 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Brian, I have been making efforts to fully support FreeSWITCH in AstLinux. Our primary targets are low powered x86 boards like the Soekris and Alix. x86, powerful enough, cheap enough (as low as $100), and about 12 watts. Not bad. The Soekris net5501 and standard case will (I believe) take a full height card. Then again you could use any board and get an external SIP gateway (ATA). We don't currently support OpenZAP with FS in AstLinux but I'd love to add support for it eventually. I'm currently working with the FS devs on getting some issues in trunk resolved to get cross compiling working again. Until then you can find ISOs with FreeSWITCH and AstLInux here if you'd like to check it out: http://mirror.astlinux.org/freeswitch/daily/ Let me know what you think. On Wed, Dec 9, 2009 at 7:55 PM, Brian May wrote: > Hello, > > I asked this question on my local linux user group mailing list, and got the > recommendation to ask here. > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > http://www.rowetel.com/ucasterisk/ip04.html > > It works well most of the time, however there are some bugs that do, under > circumstances lead to less then desirable behaviour (such as on some occasions > which I don't fully understand sometimes the remote system fails to generate > any audio packets when there is no audio - almost like silence suppression was > supported by the remote system - and asterisk fails to generate any audio > packets in return; on another slower computer running the same SIP software and > on the same network everything works fine; as far as I can tell the software - > twinkle - doesn't even support silence suppression). > > I suspect at least some - if not all - of the issues I have encountered may be > resolved with Freeswitch, however I don't really want to replace my small, > energy efficient, embedded system, with a large, power hungry computer system. > Overkill. > > An added complication is I need at least 1 analogue port to connect to the > Australian based telephone line (2 ports exchange ports and 1 extension port > would be ideal but not essiential). > > Unfortunately, I have been told that the IP04 hardware isn't compatable with > the requirements of Freeswitch. Such as not having a MMU. So there doesn't > appear to be much effort porting Freeswitch to IP04 as a result. > > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. > > Are there any other good alternatives? > > Thanks. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From frank at carmickle.com Wed Dec 9 19:50:06 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 9 Dec 2009 22:50:06 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <20091210035006.GN31924@base.carmickle.com> On Thu, Dec 10, Brian May wrote: > Hello, > > I asked this question on my local linux user group mailing list, and got the > recommendation to ask here. > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > http://www.rowetel.com/ucasterisk/ip04.html > > It works well most of the time, however there are some bugs that do, under > circumstances lead to less then desirable behaviour (such as on some occasions > which I don't fully understand sometimes the remote system fails to generate > any audio packets when there is no audio - almost like silence suppression was > supported by the remote system - and asterisk fails to generate any audio > packets in return; on another slower computer running the same SIP software and > on the same network everything works fine; as far as I can tell the software - > twinkle - doesn't even support silence suppression). > > I suspect at least some - if not all - of the issues I have encountered may be > resolved with Freeswitch, however I don't really want to replace my small, > energy efficient, embedded system, with a large, power hungry computer system. > Overkill. > > An added complication is I need at least 1 analogue port to connect to the > Australian based telephone line (2 ports exchange ports and 1 extension port > would be ideal but not essiential). > > Unfortunately, I have been told that the IP04 hardware isn't compatable with > the requirements of Freeswitch. Such as not having a MMU. So there doesn't > appear to be much effort porting Freeswitch to IP04 as a result. > > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. > > Are there any other good alternatives? A board with an atom 330 on it would probably do the trick for you. There are a few made by Intel and Supermicro that look pretty nice. There were some other people on the list looking to use them. Maybe we can get a report from someone. --FC From jason at jasonjgw.net Wed Dec 9 19:55:29 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 10 Dec 2009 14:55:29 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210005545.GD28041@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> Message-ID: <20091210035529.GA23724@jdc.jasonjgw.net> Brian May wrote: > I do have a spare TDM400p card, although as it is full height, suspect this > isn't going to help. Have a look at http://www.yawarra.com.au/ Some of their hardware (notably the Soekris Engineering boards: http://www.soekris.com/) has a PCI slot. Disclaimer: in principle this should work well with FreeSWITCH, but I haven't tested it as I don't own the hardware yet. From mouncifbb at gmail.com Wed Dec 9 20:06:02 2009 From: mouncifbb at gmail.com (Mouncif Benniane) Date: Wed, 9 Dec 2009 23:06:02 -0500 Subject: [Freeswitch-users] Generate cdrs In-Reply-To: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> References: <191c3a030912041548jb74afb7id97d341fab7149a1@mail.gmail.com> Message-ID: how big does need to get before it rotates, what's the size exactly? also how do I do it through dialplan via javascript? On Fri, Dec 4, 2009 at 6:48 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > set rotate-on-hup to false in the cdr_csv config file > then it will only rotate when the file gets too big > > and also you can get a cdr with > > session.generateXmlCdr() and dig out what you need or get it from > variables but it will not be nearly as reliable as using the C ones because > you need low level access to make sure you write to the disk properly from > many threads etc. > > > On Thu, Dec 3, 2009 at 4:33 PM, Mouncif Benniane wrote: > >> is it possible to run a javascript at the end of dialplan to generate >> cdrs? because (mod_cdr_csv) is giving me hard time as it rotates Master file >> on machine reboots or shutdown signals. >> javascript or LUA for preferences? >> >> thank you >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091209/700ffd8d/attachment-0002.html From brian at microcomaustralia.com.au Wed Dec 9 20:53:32 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 15:53:32 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <4B207ECC.1020405@microcomaustralia.com.au> Kristian Kielhofner wrote: > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > Ok, I found this: . It looks like room for a full height card. 4 network adaptors for a Freeswitch box. Hmmm. Suspect I would only find use for one ;-) Lack of OpenZAP support might be an issue, I assume that would be required to connect to an onboard analogue port... I assume I could just install Debian or another distribution instead though. Does this require a hard disk drive to boot Linux? I am guessing that compact flash could be used instead. Alternatively, if I used an external ATA, what is a good one to use? I think Jason has already made a suggestion, if so I have forgotten. I guess I get nervous going down this approach because it will add to the latency, but then again it won't use so much CPU power either, and the Digium cards send a lot of time-critical interrupts. > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: I am curious, how do you install ISOs onto a box like the net5501? I don't see any provision for CD-ROM drives. -- Brian May From brian at microcomaustralia.com.au Wed Dec 9 21:21:20 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Thu, 10 Dec 2009 16:21:20 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210035529.GA23724@jdc.jasonjgw.net> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035529.GA23724@jdc.jasonjgw.net> Message-ID: <4B208550.40104@microcomaustralia.com.au> Jason White wrote: > Have a look at http://www.yawarra.com.au/ > Ok, found the net5501: http://www.yawarra.com.au/hw-net5501.php And here it is assembled for you: http://www.yawarra.com.au/product.php?productCode=HW-NT55 I am not quite sure on one aspect, for extensions to work the TDM400P card requires a IDE style power connector that provides 12V, 5V, etc. Presumably this would be possible somehow with the net5501, because those voltages would be required for a HDD which seems to be supported. Anyone know what are the "Pigtail" and "DIN rail clips" options? -- Brian May From mike at jerris.com Wed Dec 9 23:18:00 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 02:18:00 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <3DB09BC7-3829-43B5-93A0-A1580768C7B0@jerris.com> I think I fixed the spandsp cross compile issues tonight, but I suspect there is a good chance that I broke other builds in the process. I also did a bunch of work to make the OS X Snow Leopard build cleaner today. Testing would be much appreciated on both. Mike On Dec 9, 2009, at 10:47 PM, Kristian Kielhofner wrote: > Brian, > > I have been making efforts to fully support FreeSWITCH in AstLinux. > Our primary targets are low powered x86 boards like the Soekris and > Alix. x86, powerful enough, cheap enough (as low as $100), and about > 12 watts. Not bad. > > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: > > http://mirror.astlinux.org/freeswitch/daily/ > > Let me know what you think. > > On Wed, Dec 9, 2009 at 7:55 PM, Brian May > wrote: >> Hello, >> >> I asked this question on my local linux user group mailing list, and got the >> recommendation to ask here. >> >> Anyway, at the moment I am running Asterisk on an IP04 embedded system. >> http://www.rowetel.com/ucasterisk/ip04.html >> >> It works well most of the time, however there are some bugs that do, under >> circumstances lead to less then desirable behaviour (such as on some occasions >> which I don't fully understand sometimes the remote system fails to generate >> any audio packets when there is no audio - almost like silence suppression was >> supported by the remote system - and asterisk fails to generate any audio >> packets in return; on another slower computer running the same SIP software and >> on the same network everything works fine; as far as I can tell the software - >> twinkle - doesn't even support silence suppression). >> >> I suspect at least some - if not all - of the issues I have encountered may be >> resolved with Freeswitch, however I don't really want to replace my small, >> energy efficient, embedded system, with a large, power hungry computer system. >> Overkill. >> >> An added complication is I need at least 1 analogue port to connect to the >> Australian based telephone line (2 ports exchange ports and 1 extension port >> would be ideal but not essiential). >> >> Unfortunately, I have been told that the IP04 hardware isn't compatable with >> the requirements of Freeswitch. Such as not having a MMU. So there doesn't >> appear to be much effort porting Freeswitch to IP04 as a result. >> >> I do have a spare TDM400p card, although as it is full height, suspect this >> isn't going to help. >> >> Are there any other good alternatives? >> >> Thanks. >> -- >> Brian May >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tculjaga at gmail.com Thu Dec 10 00:05:56 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 10 Dec 2009 09:05:56 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> Message-ID: <65d96fc80912100005p5be13c65p50431025d882034a@mail.gmail.com> Kristian, from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is the maximum simultaneous calls that this box can handle of course using g729 codec? I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7 simultaneous calls. To be honest, i didnt run AstLinux on alix i used voyage instead but anyhow... this seems to be the limit. what i'm looking for it an appliance to run 2-16 FXS on it.... any suggestion? T. On Thu, Dec 10, 2009 at 4:47 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Brian, > > I have been making efforts to fully support FreeSWITCH in AstLinux. > Our primary targets are low powered x86 boards like the Soekris and > Alix. x86, powerful enough, cheap enough (as low as $100), and about > 12 watts. Not bad. > > The Soekris net5501 and standard case will (I believe) take a full > height card. Then again you could use any board and get an external > SIP gateway (ATA). We don't currently support OpenZAP with FS in > AstLinux but I'd love to add support for it eventually. > > I'm currently working with the FS devs on getting some issues in > trunk resolved to get cross compiling working again. Until then you > can find ISOs with FreeSWITCH and AstLInux here if you'd like to check > it out: > > http://mirror.astlinux.org/freeswitch/daily/ > > Let me know what you think. > > On Wed, Dec 9, 2009 at 7:55 PM, Brian May > wrote: > > Hello, > > > > I asked this question on my local linux user group mailing list, and got > the > > recommendation to ask here. > > > > Anyway, at the moment I am running Asterisk on an IP04 embedded system. > > http://www.rowetel.com/ucasterisk/ip04.html > > > > It works well most of the time, however there are some bugs that do, > under > > circumstances lead to less then desirable behaviour (such as on some > occasions > > which I don't fully understand sometimes the remote system fails to > generate > > any audio packets when there is no audio - almost like silence > suppression was > > supported by the remote system - and asterisk fails to generate any audio > > packets in return; on another slower computer running the same SIP > software and > > on the same network everything works fine; as far as I can tell the > software - > > twinkle - doesn't even support silence suppression). > > > > I suspect at least some - if not all - of the issues I have encountered > may be > > resolved with Freeswitch, however I don't really want to replace my > small, > > energy efficient, embedded system, with a large, power hungry computer > system. > > Overkill. > > > > An added complication is I need at least 1 analogue port to connect to > the > > Australian based telephone line (2 ports exchange ports and 1 extension > port > > would be ideal but not essiential). > > > > Unfortunately, I have been told that the IP04 hardware isn't compatable > with > > the requirements of Freeswitch. Such as not having a MMU. So there > doesn't > > appear to be much effort porting Freeswitch to IP04 as a result. > > > > I do have a spare TDM400p card, although as it is full height, suspect > this > > isn't going to help. > > > > Are there any other good alternatives? > > > > Thanks. > > -- > > Brian May > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/f722c55c/attachment-0002.html From yehavi.bourvine at gmail.com Thu Dec 10 01:11:51 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 10 Dec 2009 11:11:51 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: An intermediate report: *Audiocodes*: TLS works only on outgoing requests, incoming ones are ignored. I am waiting for Audiocodes' help in order to debug it. SRTP: worked when no TLS is active. When TLS is active the call is disconnected when the remote party answers. Still debugging it. *VegaStream Europa-50*: SRTP works. Waiting for Vega for instructions how to enable TLS from the WEB interface. Regards, __Yehavi: 2009/12/4 Yehavi Bourvine > I'll report when I am done. > > So far I've enabled only SRTP and both support it. > > __Yehavi: > > 2009/12/4 Mark Campbell-Smith > >> Thanks Yehavi, >> >> I would be very interested to find out how your test goes... can you >> report back after you have tested it? >> >> Thanks! >> >> On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine >> wrote: >> > Hello, >> > >> > I have AudioCodes MP and Vega ATA adapters. They both support SRTP; >> they >> > should support TLS also (will try it next week; up to now I preffered to >> not >> > use TLS so I can sniff the traffic and debug things). >> > >> > Regards, __Yehavi: >> > >> > 2009/12/4 Mark Campbell-Smith >> >> >> >> Cheers Gabriel.. thanks for the information. >> >> >> >> I'll look at the Mediatrix ATA's as an alternative - has anyone had >> >> experience with those and TLS/SRTP? >> >> >> >> >> >> On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri wrote: >> >> > The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are >> the >> >> > Grandstream and Mediatrix devices (although I've never tried either >> >> > one with FreeSWITCH). >> >> > >> >> > I've personally never had any good experience with the Grandstream >> >> > ATAs. The Mediatrix ATAs are OK devices, but I've never personally >> >> > tested them with SRTP w/SDES and FreeSWITCH, but supposedly they >> >> > support it (so says their marketing material and docs). >> >> > >> >> > I'd see if Cisco has any plans to add support for it to the ATAs. >> Next >> >> > time I see our Cisco SE, I'll try to poke him about it. >> >> > >> >> > Gabe >> >> > >> >> > On Thu, Dec 3, 2009 at 2:34 PM, Mark Campbell-Smith >> >> > wrote: >> >> >> Quote: Cisco/Linksys SPA series ATAs do not support SDES key >> exchange >> >> >> to appropriately support SRTP and FreeSWITCH >> >> >> >> >> >> I'll check with Cisco regarding their implementation then and try to >> >> >> find out when/if they will support standard SRTP encryption. >> >> >> >> >> >> >> >> >> So, back to my origianal question then. Are there any ATA's that >> >> >> support TLS AND SRTP with FreeSwitch? >> >> >> >> >> >> >> >> >> On Fri, Dec 4, 2009 at 9:17 AM, Gabriel Kuri >> wrote: >> >> >>> AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key >> >> >>> exchange to appropriately support SRTP and FreeSWITCH. They do >> their >> >> >>> proprietary Sipura key exchange only, not sure if Cisco plans on >> >> >>> upgrading the firmware to ever support SDES on the ATAs. They added >> >> >>> support for SDES to their IP Phones about 1 year ago, but nothing >> has >> >> >>> happened with the ATAs as of yet. >> >> >>> >> >> >>> Gabe >> >> >>> >> >> >>> >> >> >>> On Thu, Dec 3, 2009 at 2:05 PM, Mark Campbell-Smith >> >> >>> wrote: >> >> >>>> Hi All, >> >> >>>> >> >> >>>> I managed to borrow a SPA3102 with the latest firmware and have >> got >> >> >>>> it >> >> >>>> to register using TLS, but I am still struggling with SRTP. Has >> >> >>>> anyone managed to get SRTP working with the Linksys devices and if >> >> >>>> so, >> >> >>>> can they direct me on how to do this. >> >> >>>> >> >> >>>> I have generated a mini-certificates and SRTP Private Key using >> the >> >> >>>> gen-mc tool found at >> >> >>>> >> >> >>>> >> http://www.megajournal.ru/journal/users_data/11049/msg_files/24120/gen-mc.c-v0.98.tar.gz.mp3 >> . >> >> >>>> However, when ever I initiate a call from the SPA, I can see that >> >> >>>> the >> >> >>>> call is not encrypted. >> >> >>>> >> >> >>>> Help appreciated. >> >> >>>> >> >> >>>> Thanks! >> >> >>>> >> >> >>>> >> >> >>>> On Sat, Nov 28, 2009 at 6:31 AM, eman wrote: >> >> >>>>> Check out the Linksys SPA2102 >> >> >>>>> >> >> >>>>> On Wed, Nov 25, 2009 at 3:34 AM, Mark Campbell-Smith >> >> >>>>> wrote: >> >> >>>>>> >> >> >>>>>> The only ATA mentioned on the WIKI that supports TLS/SRTP is the >> >> >>>>>> Grandstream HandyTone 503. But, again according to the wiki, >> that >> >> >>>>>> doesn't seem to behave to well with TLS ... >> >> >>>>>> >> >> >>>>>> On Wed, Nov 25, 2009 at 7:14 PM, Jason White < >> jason at jasonjgw.net> >> >> >>>>>> wrote: >> >> >>>>>> > Mark Campbell-Smith wrote: >> >> >>>>>> >> Does the SPA3102 support TLS or only SRTP? >> >> >>>>>> > >> >> >>>>>> > I don't know, but supporting only SRTP would be ridiculous, >> since >> >> >>>>>> > the >> >> >>>>>> > keys >> >> >>>>>> > would then be transmitted in the clear and therefore amenable >> to >> >> >>>>>> > interception. >> >> >>>>>> > SRTP requires the SIP channel to be encrypted by TLS in order >> to >> >> >>>>>> > be >> >> >>>>>> > secure. >> >> >>>>>> > ZRTP, on the other hand, doesn't have this limitation: it >> works >> >> >>>>>> > entirely >> >> >>>>>> > in >> >> >>>>>> > RTP. >> >> >>>>>> > >> >> >>>>>> > I would be rather surprised were a hardware manufacturer to >> >> >>>>>> > implement >> >> >>>>>> > SRTP >> >> >>>>>> > without TLS for the SIP traffic. On the other hand, we've seen >> >> >>>>>> > often in >> >> >>>>>> > this >> >> >>>>>> > forum that some manufacturers are really clueless... >> >> >>>>>> > >> >> >>>>>> > >> >> >>>>>> > _______________________________________________ >> >> >>>>>> > FreeSWITCH-users mailing list >> >> >>>>>> > FreeSWITCH-users at lists.freeswitch.org >> >> >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>>> > >> >> >>>>>> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> > http://www.freeswitch.org >> >> >>>>>> > >> >> >>>>>> >> >> >>>>>> _______________________________________________ >> >> >>>>>> FreeSWITCH-users mailing list >> >> >>>>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>>> >> >> >>>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> http://www.freeswitch.org >> >> >>>>> >> >> >>>>> >> >> >>>>> _______________________________________________ >> >> >>>>> FreeSWITCH-users mailing list >> >> >>>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>>> >> >> >>>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >> >> >>>>> >> >> >>>>> >> >> >>>> >> >> >>>> _______________________________________________ >> >> >>>> FreeSWITCH-users mailing list >> >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>>> >> >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>> http://www.freeswitch.org >> >> >>>> >> >> >>> >> >> >>> _______________________________________________ >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> _______________________________________________ >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> > _______________________________________________ >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/0046d3cb/attachment-0002.html From asterisk at dotr.com Thu Dec 10 02:13:16 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Thu, 10 Dec 2009 10:13:16 +0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: Sometime next week I hopefully am going to start a document that follows my progress in setting up a FS system from scratch, with all the pitfalls and successes. A kinds of "warts and all" story. Alongside this "blog" (for want of a better word) I will also then document the steps needed to get it working (a howto guide, effectively). I am a long time * user (2004), so my mindset is kind of skewed - but perhaps that would be beneficial for other * users looking at implementing FS. Most of our config and dialplan is generated by using res_config_curl, and we use things like call listening, conferencing, parking and queues. We do use queues in a slightly odd manner (we add 1 agent, and call a local channel). When this channels is called, we use curl to get our application to return the most appropriate agent to actually call). We also use * as a power dialler, making upwards of 400,000 call attempts per month. Not massive, but not tiny either. Hopefully, this will be of use to both FS and * users. What would be great is that if other people follow my progress, and make suggestions as and when I hit a brick wall :) What would be best for this ? A blog ? Or a wiki page ? Julian 2009/12/9 Brian West : > That is what is nice about our community I'm more than willing to answer the > questions if you document them... as are many others in the core team...we > just have a lot to do and I think the best repayment is documentation! ;) > /b > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > > On Thu, Dec 10, 2009 at 11:07 AM, Brian West wrote: > > Visit the friday meetings and we can help if you document it. ?;) > > > I would be willing to lend a hand with the documentation but I know so > little (a complete freeswitch noob). For example I was trying to > figure out how to tell if an extension was set up "show dialplan in > asterisk". ?I could not find this anywhere. If I find out I would be > happy to add it to the rosetta stone. > > I am currently working on getting outbound socket working. Once I get > it going I would be happy to add it to the relevant section of the > wiki (in this case ruby). > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From codecomplete at free.fr Thu Dec 10 03:40:11 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 03:40:11 -0800 (PST) Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> Message-ID: <26725860.post@talk.nabble.com> No publisher, although uploading and selling books (deadtree or online) is easy with companies like www.lulu.com I was just thinking of some way to learn FS gradually and effectively. The frequent problem with wiki's, is that the quality of articles is uneven and they don't have a good layout. But then, writing documentation is hard and time-consuming :-/ -- View this message in context: http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p26725860.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 03:45:09 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 03:45:09 -0800 (PST) Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210035006.GN31924@base.carmickle.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> Message-ID: <26725918.post@talk.nabble.com> Frank Carmickle wrote: > A board with an atom 330 on it would probably do the trick for you. There > are a few made by Intel and Supermicro that look pretty nice. There were > some other people on the list looking to use them. Maybe we can get a > report from someone. Intel came up with the D945GSEJT, which is totally fanless and has an embedded DC/DC, so all you have to add is an external AC/DC power brick, some RAM, a PCI riser to save space, and either a hard-disk or a CompactFlash + IDE adaptor. I'm thinking of building one with a Digium-compatible PCI card. www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm -- View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mcampbellsmith at gmail.com Thu Dec 10 04:13:07 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 10 Dec 2009 23:13:07 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <26725918.post@talk.nabble.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> Message-ID: <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> I use a dectop by Data Evolution... Its cheap at ~$100. I have it running debian lenny and FS... works well for me. http://www.dataevolution.com/dectop%20info%202.htm http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/ On Thu, Dec 10, 2009 at 10:45 PM, Fred-145 wrote: > > > Frank Carmickle wrote: >> A board with an atom 330 on it would probably do the trick for you. ?There >> are a few made by Intel and Supermicro that look pretty nice. ?There were >> some other people on the list looking to use them. ?Maybe we can get a >> report from someone. > > Intel came up with the D945GSEJT, which is totally fanless and has an > embedded DC/DC, so all you have to add is an external AC/DC power brick, > some RAM, a PCI riser to save space, and either a hard-disk or a > CompactFlash + IDE adaptor. I'm thinking of building one with a > Digium-compatible PCI card. > > www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm > -- > View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26725918.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From codecomplete at free.fr Thu Dec 10 04:57:21 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 04:57:21 -0800 (PST) Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <33c87fa30912100413r5703fa45w32d148c39a8af841@mail.gmail.com> Message-ID: <26726783.post@talk.nabble.com> Mark Campbell-Smith wrote: > I use a dectop by Data Evolution... Its cheap at ~$100. I have it > running debian lenny and FS... works well for me. Thanks for the tip, although this type of box doesn't have a PCI slot, so the only way to connect FS to the PSTN is through a VoIP provider (or a Linksys 3102, with the usual, possible echo issues). In the same vein as the decTop : http://en.wikipedia.org/wiki/SheevaPlug Since FS can be compiled to ARM, Freeswitch might be able to run on this device: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux -- View this message in context: http://old.nabble.com/embedded-freeswitch-compatable-hardware-tp26721589p26726783.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 05:40:03 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 05:40:03 -0800 (PST) Subject: [Freeswitch-users] [vars.xml] default_password=1234? Message-ID: <26727371.post@talk.nabble.com> Hello I'm going through the various XML files, and noticed this first line in vars.xml. What is this password used for? Thank you -- View this message in context: http://old.nabble.com/-vars.xml--default_password%3D1234--tp26727371p26727371.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 10 05:45:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 07:45:28 -0600 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <164a9ab00911271131i1e8052e0uac0f4471e4e21733@mail.gmail.com> <33c87fa30912031405t59554fc0l10392a34b113f18@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: I have confirmed it works with Polycom, Snom and a few others .... polycom is the hardest to set due to having to put the ca cert into the phone... but other than that its good. /b On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > An intermediate report: > > Audiocodes: TLS works only on outgoing requests, incoming ones are > ignored. I am waiting for Audiocodes' help in order to debug it. > SRTP: worked when no TLS is active. When TLS is active the call is > disconnected when the remote party answers. Still debugging it. > > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > how to enable TLS from the WEB interface. > > Regards, __Yehavi: From brian at freeswitch.org Thu Dec 10 05:47:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 07:47:28 -0600 Subject: [Freeswitch-users] [vars.xml] default_password=1234? In-Reply-To: <26727371.post@talk.nabble.com> References: <26727371.post@talk.nabble.com> Message-ID: please look in conf/directory/default/*.xml /b On Dec 10, 2009, at 7:40 AM, Fred-145 wrote: > > Hello > > I'm going through the various XML files, and noticed this first line > in > vars.xml. > > > > What is this password used for? > > Thank you From codecomplete at free.fr Thu Dec 10 06:04:38 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 06:04:38 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? Message-ID: <26727762.post@talk.nabble.com> Hello I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP activity" option unchecked, ran FreeSwitch, and used eg. Shields Up (www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened... which SU says no. Does it mean that... - by default, FS doesn't use STUN - or my modem doesn't support STUN, and I must either enable UPnP or map ports (UDP5080 and some UDP ports for RTP/RTPC) statically? Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26727762.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Thu Dec 10 06:09:17 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 06:09:17 -0800 (PST) Subject: [Freeswitch-users] [vars.xml] default_password=1234? In-Reply-To: References: <26727371.post@talk.nabble.com> Message-ID: <26727835.post@talk.nabble.com> Ah, makes sense: conf/directory/default/1000.xml: Thanks for the tip. -- View this message in context: http://old.nabble.com/-vars.xml--default_password%3D1234--tp26727371p26727835.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Dec 10 06:26:17 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 10 Dec 2009 15:26:17 +0100 Subject: [Freeswitch-users] Invite local number into a conference - codec problem Message-ID: <4B210509.4000004@gmx.net> Hello, I try to invite a user into a conference by loopback/255 8000 Conference 255 is the user, I invite the user via loopback as that way I can also invite external numbers. It processes the user's local dialplan correctly (as if the user was normally dialled), however it only offers L16 codec, so the Phone fails. I can see no codec negociation on the debug console. If I call the phone from another phone, then codec negociation is taking place. If I invite an external PSTN user into the conference then codecs are set correctly (L16+PCMA+PCMU etc) Is there a way to explicitely set the codec for the conference? is not set is, still commented in the internal profile. In vars.conf.xml only only PCMA and PCMU are set. Best regards Peter From frank at carmickle.com Thu Dec 10 06:26:58 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 10 Dec 2009 09:26:58 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <26725918.post@talk.nabble.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> Message-ID: <20091210142658.GP31924@base.carmickle.com> On Thu, Dec 10, Fred-145 wrote: > > > Frank Carmickle wrote: > > A board with an atom 330 on it would probably do the trick for you. There > > are a few made by Intel and Supermicro that look pretty nice. There were > > some other people on the list looking to use them. Maybe we can get a > > report from someone. > > Intel came up with the D945GSEJT, which is totally fanless and has an > embedded DC/DC, so all you have to add is an external AC/DC power brick, > some RAM, a PCI riser to save space, and either a hard-disk or a > CompactFlash + IDE adaptor. I'm thinking of building one with a > Digium-compatible PCI card. > > www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm The 330 boards are a little more power hungry but you get a dual core 64 bit processor. As far as I'm concerned the performance increase is well worth the extra money. You still well below the power consumption of any other 64 bit dual core machines. http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 --FC From rupa at rupa.com Thu Dec 10 06:33:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 10 Dec 2009 08:33:58 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26727762.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> Message-ID: STUN is not a way to open ports in a manner in which sheilds up would detect. http://en.wikipedia.org/wiki/STUN UPNP is what you want if you want to open ports. STUN is just a method for figuring out how to do nat traversal. STUN "method" is initiated by the process on the inside of the firewall at connection establish time. It will do no good for the listen case which is what you are checking for. If you want to use stun, then you need to forward the listen ports manually (5060, 5080 - UDP and TCP). On Thu, Dec 10, 2009 at 8:04 AM, Fred-145 wrote: > > Hello > > I wanted to check if my ADSL modem worked with STUN, so I left its "UPNP > activity" option unchecked, ran FreeSwitch, and used eg. Shields Up > (www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened... > which SU says no. > > Does it mean that... > - by default, FS doesn't use STUN > - or my modem doesn't support STUN, and I must either enable UPnP or map > ports (UDP5080 and some UDP ports for RTP/RTPC) statically? > > Thank you. > -- > View this message in context: > http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26727762.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/670cff3f/attachment-0002.html From kristian.kielhofner at gmail.com Thu Dec 10 08:12:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 10 Dec 2009 11:12:07 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091210142658.GP31924@base.carmickle.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <20091210142658.GP31924@base.carmickle.com> Message-ID: <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle wrote: > > The 330 boards are a little more power hungry but you get a dual core 64 bit processor. ?As far as I'm concerned the performance increase is well worth the extra money. ?You still well below the power consumption of any other 64 bit dual core machines. > > http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 > > --FC > While these are low power when compared to traditional desktop/server systems, they're not what I would consider to be embedded. The CPU requires a fan (embedded no-no) and between the chipset and CPU they draw several times more power than a traditional embedded system. The ALIX and Soekris boards run with 12 watt power supplies (12v, 1 amp). The Atom 330 alone can draw 8 watts. This is still impressive for a processor of this class but it's not what I would consider to be embedded, yet... I think of embedded systems like this: Blackfin - Very low power, good performance (especially for DSP), very difficult porting (usually) ARM/MIPS - Very low power, decent performance depending on application, mild difficulty in porting X86 (Geode, etc) - Pretty low power, decent performance, relative ease in "porting" (often none) Everything else - You should probably call it an "appliance", not an embedded system With the correct target application and design ARM and Geode systems can provide more than enough CPU power for many, many practical applications. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Thu Dec 10 08:35:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 10:35:22 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: <191c3a030912100835p1cba7c5m5abca73d2ecfcc58@mail.gmail.com> Don't worry. I was an asterisk developer/volunteer in 2003. I still managed to figure it out. ;) On Thu, Dec 10, 2009 at 4:13 AM, Julian Lyndon-Smith wrote: > Sometime next week I hopefully am going to start a document that > follows my progress in setting up a FS system from scratch, with all > the pitfalls and successes. A kinds of "warts and all" story. > Alongside this "blog" (for want of a better word) I will also then > document the steps needed to get it working (a howto guide, > effectively). > > I am a long time * user (2004), so my mindset is kind of skewed - but > perhaps that would be beneficial for other * users looking at > implementing FS. > > Most of our config and dialplan is generated by using res_config_curl, > and we use things like call listening, conferencing, parking and > queues. We do use queues in a slightly odd manner (we add 1 agent, and > call a local channel). When this channels is called, we use curl to > get our application to return the most appropriate agent to actually > call). > > We also use * as a power dialler, making upwards of 400,000 call > attempts per month. Not massive, but not tiny either. > > Hopefully, this will be of use to both FS and * users. What would be > great is that if other people follow my progress, and make suggestions > as and when I hit a brick wall :) > > What would be best for this ? A blog ? Or a wiki page ? > > Julian > > 2009/12/9 Brian West : > > That is what is nice about our community I'm more than willing to answer > the > > questions if you document them... as are many others in the core > team...we > > just have a lot to do and I think the best repayment is documentation! ;) > > /b > > On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote: > > > > On Thu, Dec 10, 2009 at 11:07 AM, Brian West > wrote: > > > > Visit the friday meetings and we can help if you document it. ;) > > > > > > I would be willing to lend a hand with the documentation but I know so > > little (a complete freeswitch noob). For example I was trying to > > figure out how to tell if an extension was set up "show dialplan in > > asterisk". I could not find this anywhere. If I find out I would be > > happy to add it to the rosetta stone. > > > > I am currently working on getting outbound socket working. Once I get > > it going I would be happy to add it to the relevant section of the > > wiki (in this case ruby). > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/18fbd093/attachment-0002.html From abeka at greatiam.com Thu Dec 10 08:42:14 2009 From: abeka at greatiam.com (Otis) Date: Thu, 10 Dec 2009 16:42:14 +0000 Subject: [Freeswitch-users] Routing calls to Another FS server Message-ID: <4B2124E6.6010507@greatiam.com> I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set as gateways and register with each other. I wanted all users on FS1 to dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as 331001 on FS1. I have a dialplan as follows in .../dialplan/default/ callweaver.xml > > > > > I have also used I have also used the line in place of without any joy. I am getting error Not - found from the client. I am registered as 1001 on FS1. Please how do I make all users use this dial plan and may I know which version of all those stated above is right. All are in the ...dialplan/default directory. called callweaver.xml Should it have a particular name either than the gateway name ? Thanks for your time once again From anthony.minessale at gmail.com Thu Dec 10 08:53:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 10:53:44 -0600 Subject: [Freeswitch-users] Invite local number into a conference - codec problem In-Reply-To: <4B210509.4000004@gmx.net> References: <4B210509.4000004@gmx.net> Message-ID: <191c3a030912100853j3eea291bwd10d3e82b43e6dd8@mail.gmail.com> set absolute_codec_string to whatever codec you want to offer in the {} on the bridge string On Thu, Dec 10, 2009 at 8:26 AM, Peter P GMX wrote: > Hello, > > I try to invite a user into a conference by > loopback/255 8000 Conference > 255 is the user, I invite the user via loopback as that way I can also > invite external numbers. > > It processes the user's local dialplan correctly (as if the user was > normally dialled), however it only offers L16 codec, so the Phone fails. > I can see no codec negociation on the debug console. > If I call the phone from another phone, then codec negociation is taking > place. > If I invite an external PSTN user into the conference then codecs are > set correctly (L16+PCMA+PCMU etc) > > Is there a way to explicitely set the codec for the conference? > > is not set is, > still commented in the internal profile. > In vars.conf.xml only only PCMA and PCMU are set. > > Best regards > Peter > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/d6211aea/attachment-0002.html From abeka at greatiam.com Thu Dec 10 09:03:40 2009 From: abeka at greatiam.com (Otis) Date: Thu, 10 Dec 2009 17:03:40 +0000 Subject: [Freeswitch-users] Routing calls to Another FS server In-Reply-To: <4B2124E6.6010507@greatiam.com> References: <4B2124E6.6010507@greatiam.com> Message-ID: <4B2129EC.6000607@greatiam.com> Otis wrote: >
I have 2 > FS servers FS1 (aka medion) and FS3 (callweaver). These are set as > gateways and register with each other. I wanted all users on FS1 to > dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed > as 331001 on FS1. > > I have a dialplan as follows in .../dialplan/default/ callweaver.xml > >> >> >> >> >> > > I have also used > > > > data="sofia/profilename/$1 at 192.168.1.110"/> > > > > I have also used the line data="sofia/gateway/outbound.callweaver/$1"/> > in place of data="sofia/profilename/$1 at 192.168.1.110"/> > without any joy. > > I am getting error Not - found from the client. I am registered as > 1001 on FS1. > > Please how do I make all users use this dial plan and may I know > which version of all those stated above is right. All are in the > ...dialplan/default directory. called callweaver.xml > > Should it have a particular name either than the gateway name ? > > Thanks for your time once again > > > >
> Sorry I forgot to add that where it says profilename I have *callweaver* which is the profile name of the gateway in /conf/sip_profiles/external/callweaver.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/032209e8/attachment-0002.html From aep.lists at it46.se Thu Dec 10 09:07:30 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 10 Dec 2009 18:07:30 +0100 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR Message-ID: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> Hi, I am currently creating IVR using the functions provided in the XML dialplan http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr Using functions like this I can play files, etc. I wonder what is the smartest way to monitor (as in big brother) the options selected by the user: I assume that I can include an entry of the type: and include in foo.js the code to track the selection. But I wonder if this is the best approach /aep -- Stopping junk mailers is good for the environment From codecomplete at free.fr Thu Dec 10 09:21:43 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 10 Dec 2009 09:21:43 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: References: <26727762.post@talk.nabble.com> Message-ID: <26731188.post@talk.nabble.com> Thanks for the clarification. So it's either UPnP or STUN/port-mapping. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26731188.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mailinglist at fribert.dk Thu Dec 10 09:27:19 2009 From: mailinglist at fribert.dk (mailinglist) Date: Thu, 10 Dec 2009 18:27:19 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <110796.60596.qm@web56401.mail.re3.yahoo.com> References: <4B2030A1020000E10000031D@mail.fribert.dk> <110796.60596.qm@web56401.mail.re3.yahoo.com> Message-ID: <4B213D87020000E100000322@mail.fribert.dk> Hi Mark The extensions are located under directory/default, and they look like this: As I understand them, the context set there, is the right one? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/57ee336e/attachment-0002.html From msc at freeswitch.org Thu Dec 10 09:51:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 09:51:55 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <855e4dcf0912091356j21222bdbk18fa6a8e419f3620@mail.gmail.com> <06386070-A060-44A3-89C3-2EADC60ED5DC@freeswitch.org> <855e4dcf0912091439p75880d31t1ec8139ac8b10daf@mail.gmail.com> <22FAFD82-8D61-42C8-B947-68854F32E2C7@freeswitch.org> Message-ID: <87f2f3b90912100951h74f3391ay91b354937417b741@mail.gmail.com> On Thu, Dec 10, 2009 at 2:13 AM, Julian Lyndon-Smith wrote: > Sometime next week I hopefully am going to start a document that > follows my progress in setting up a FS system from scratch, with all > the pitfalls and successes. A kinds of "warts and all" story. > Alongside this "blog" (for want of a better word) I will also then > document the steps needed to get it working (a howto guide, > effectively). > > I am a long time * user (2004), so my mindset is kind of skewed - but > perhaps that would be beneficial for other * users looking at > implementing FS. > > Most of our config and dialplan is generated by using res_config_curl, > and we use things like call listening, conferencing, parking and > queues. We do use queues in a slightly odd manner (we add 1 agent, and > call a local channel). When this channels is called, we use curl to > get our application to return the most appropriate agent to actually > call). > > We also use * as a power dialler, making upwards of 400,000 call > attempts per month. Not massive, but not tiny either. > > Hopefully, this will be of use to both FS and * users. What would be > great is that if other people follow my progress, and make suggestions > as and when I hit a brick wall :) > > What would be best for this ? A blog ? Or a wiki page ? > Julian, First off, welcome to the FreeSWITCH fold! Thank you for your willingness not only to try things but to document them as well. I like the idea of a blog to tell the story. After you're done then you might consider a wiki page that is more of "just the facts, ma'am" on how you translated Asterisk configs to a FreeSWITCH setup. Feel free to ask myself or anyone here or on IRC for assistance. -MC (IRC: mercutioviz) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/a7acf7b1/attachment-0002.html From msc at freeswitch.org Thu Dec 10 09:53:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 09:53:16 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <26725860.post@talk.nabble.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> Message-ID: <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> On Thu, Dec 10, 2009 at 3:40 AM, Fred-145 wrote: > > No publisher, although uploading and selling books (deadtree or online) is > easy with companies like www.lulu.com > > I was just thinking of some way to learn FS gradually and effectively. The > frequent problem with wiki's, is that the quality of articles is uneven and > they don't have a good layout. But then, writing documentation is hard and > time-consuming :-/ > Amen, brothah! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/e639b372/attachment-0002.html From msc at freeswitch.org Thu Dec 10 10:13:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:13:29 -0800 Subject: [Freeswitch-users] Routing calls to Another FS server In-Reply-To: <4B2124E6.6010507@greatiam.com> References: <4B2124E6.6010507@greatiam.com> Message-ID: <87f2f3b90912101013n339e7931nc50ba91328cdcb4b@mail.gmail.com> On Thu, Dec 10, 2009 at 8:42 AM, Otis wrote: > I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set > as gateways and register with each other. I wanted all users on FS1 to > dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as > 331001 on FS1. > > I have a dialplan as follows in .../dialplan/default/ callweaver.xml > > > > > > > > > > > > > I have also used > > > > > > > > I have also used the line data="sofia/gateway/outbound.callweaver/$1"/> > in place of data="sofia/profilename/$1 at 192.168.1.110"/> > without any joy. > > I am getting error Not - found from the client. I am registered as > 1001 on FS1. > Otis, I would recommend that you turn on debugging and capture the console output. Put it in a pastebin and then respond to this thread with a link to the pastebin URL. FYI, here's a handy page that will give you some nice pointers on how to gather information when you're doing troubleshooting: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has detailed instructions on getting console output, dumping to pastebin, etc. If you want to get really fancy you can also try Tony's handy-dandy debug Perl script: libs/esl/perl/logger.pl under the FS source directory. Run the script, make your call, then hit ctrl-C. The script will upload the log to pastebin for you and tell you the URL. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/8bb3e199/attachment-0002.html From msc at freeswitch.org Thu Dec 10 10:15:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:15:56 -0800 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR In-Reply-To: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> References: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> Message-ID: <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero wrote: > > Hi, > > I am currently creating IVR using the functions provided in the XML > dialplan > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr > > Using functions like this > param="$${base_dir}/1255549537_Welcome.wav"/> > I can play files, etc. > > I wonder what is the smartest way to monitor (as in big brother) the > options selected by the user: > > I assume that I can include an entry of the type: > > and include in foo.js the code to track the selection. > > But I wonder if this is the best approach > > /aep > > Are you trying to do some sort of live monitoring as it happens (i.e. while the call is live) or do you just want a record of the digits they pressed? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/1ee57f3d/attachment-0002.html From brian at freeswitch.org Thu Dec 10 10:17:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Dec 2009 12:17:06 -0600 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> Message-ID: Use the BKW method... three to four word sentences to describe what to do... its very poetic! Or is that haiku? /b On Dec 10, 2009, at 11:53 AM, Michael Collins wrote: > > I was just thinking of some way to learn FS gradually and > effectively. The > frequent problem with wiki's, is that the quality of articles is > uneven and > they don't have a good layout. But then, writing documentation is > hard and > time-consuming :-/ > > Amen, brothah! :) > -MC From mike at jerris.com Thu Dec 10 10:22:13 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 13:22:13 -0500 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26731188.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> Message-ID: <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> we also support natpmp and static ip setting. Mike On Dec 10, 2009, at 12:21 PM, Fred-145 wrote: > > Thanks for the clarification. So it's either UPnP or STUN/port-mapping. From msc at freeswitch.org Thu Dec 10 10:26:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 10:26:52 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> Message-ID: <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> On Thu, Dec 10, 2009 at 10:17 AM, Brian West wrote: > Use the BKW method... three to four word sentences to describe what to > do... its very poetic! Or is that haiku? > > /b > Update to latest Did you type make current yet? Tony hates build skew -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/12623dcc/attachment-0002.html From shaun_clark at hotmail.com Thu Dec 10 10:26:39 2009 From: shaun_clark at hotmail.com (Shaun Clark) Date: Thu, 10 Dec 2009 10:26:39 -0800 Subject: [Freeswitch-users] Route Non-Call Data to Agent Through Queue Message-ID: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> I have an application where I would like to route both calls and other requests through the same queue to the same agents, for example the same agent might take a call and then right after that take a chat. But, the chat server we use is separate from our phone system. What I would like to do is basically route some text, i.e. "new chat chat_id_goes_here" through to the agent. Is this possible with FreeSwitch? The idea being the soft-phone would receive this text and we would write code to catch this message do the appropriate action on our CRM. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/bce228c0/attachment-0002.html From aep.lists at it46.se Thu Dec 10 11:24:12 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 10 Dec 2009 20:24:12 +0100 Subject: [Freeswitch-users] Monitoring IVR pressed-options in XML IVR In-Reply-To: <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> References: <3ea0bbee83f284a6fcb85b8f45f98e7a.squirrel@correo.nodo50.org> <87f2f3b90912101015u5287b237pa00cbe5cb0490650@mail.gmail.com> Message-ID: <770de553a9869a67b2db3a1bc5fae765.squirrel@correo.nodo50.org> I want to trigger CUSTOM events via ESL "as they navigate inside" of the IVR. The XML IVRs are generated from a GUI. The CUSTOM events need to carry - what IVR the user is navigating - what option has been selected - ideally how long they stayed listening (this can be calculated) - and when they hang the phone /aep -- Stopping junk mailers is good for the environment > On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero > wrote: > >> >> Hi, >> >> I am currently creating IVR using the functions provided in the XML >> dialplan >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr >> >> Using functions like this >> > param="$${base_dir}/1255549537_Welcome.wav"/> >> I can play files, etc. >> >> I wonder what is the smartest way to monitor (as in big brother) the >> options selected by the user: >> >> I assume that I can include an entry of the type: >> >> and include in foo.js the code to track the selection. >> >> But I wonder if this is the best approach >> >> /aep >> >> Are you trying to do some sort of live monitoring as it happens (i.e. >> while > the call is live) or do you just want a record of the digits they pressed? > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tculjaga at gmail.com Thu Dec 10 12:53:51 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 10 Dec 2009 21:53:51 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> References: <20091210005545.GD28041@sys11.in.vpac.org> <20091210035006.GN31924@base.carmickle.com> <26725918.post@talk.nabble.com> <20091210142658.GP31924@base.carmickle.com> <2d9149cd0912100812r69f50c02gf6a4f97d74f1c020@mail.gmail.com> Message-ID: <65d96fc80912101253o410ff73dv213b4c6509f882ae@mail.gmail.com> ok, but how much smultaneous calls did you get on an alix board using astlinux... for istnace, this is the question? T. On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle > wrote: > > > > The 330 boards are a little more power hungry but you get a dual core 64 > bit processor. As far as I'm concerned the performance increase is well > worth the extra money. You still well below the power consumption of any > other 64 bit dual core machines. > > > > http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 > > > > --FC > > > > While these are low power when compared to traditional > desktop/server systems, they're not what I would consider to be > embedded. The CPU requires a fan (embedded no-no) and between the > chipset and CPU they draw several times more power than a traditional > embedded system. The ALIX and Soekris boards run with 12 watt power > supplies (12v, 1 amp). The Atom 330 alone can draw 8 watts. This is > still impressive for a processor of this class but it's not what I > would consider to be embedded, yet... > > I think of embedded systems like this: > > Blackfin - Very low power, good performance (especially for DSP), very > difficult porting (usually) > ARM/MIPS - Very low power, decent performance depending on > application, mild difficulty in porting > X86 (Geode, etc) - Pretty low power, decent performance, relative ease > in "porting" (often none) > Everything else - You should probably call it an "appliance", not an > embedded system > > With the correct target application and design ARM and Geode systems > can provide more than enough CPU power for many, many practical > applications. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/b0c3804f/attachment-0002.html From dave at 3c.co.uk Thu Dec 10 13:11:20 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 10 Dec 2009 14:11:20 -0700 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> Message-ID: <1260479480.12078.18.camel@local.freepabx.com> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: > Update to latest > Did you type make current yet? > Tony hates build skew Brilliant. Michael Collins-san Shrinks all usual advice Into one Haiku. --Dave From msc at freeswitch.org Thu Dec 10 13:19:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 13:19:32 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <1260479480.12078.18.camel@local.freepabx.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> Message-ID: <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> On Thu, Dec 10, 2009 at 1:11 PM, David Knell wrote: > > On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: > > > Update to latest > > Did you type make current yet? > > Tony hates build skew > > Brilliant. > > Michael Collins-san > Shrinks all usual advice > Into one Haiku. > > --Dave > I love the wiki The docs are disorganized I hate the wiki -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/530c6846/attachment-0002.html From asterisk at dotr.com Thu Dec 10 14:16:54 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Thu, 10 Dec 2009 22:16:54 +0000 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> Message-ID: Ok. The journey begins. http://makingfs.blogspot.com/ Don't know if you want to add this link to the website or wiki. Julian 2009/12/10 Michael Collins : > > > On Thu, Dec 10, 2009 at 1:11 PM, David Knell wrote: >> >> On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote: >> >> > Update to latest >> > Did you type make current yet? >> > Tony hates build skew >> >> Brilliant. >> >> Michael Collins-san >> Shrinks all usual advice >> Into one Haiku. >> >> --Dave > > I love the wiki > The docs are disorganized > I hate the wiki > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From andrew at hijacked.us Thu Dec 10 14:32:58 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 10 Dec 2009 17:32:58 -0500 Subject: [Freeswitch-users] Route Non-Call Data to Agent Through Queue In-Reply-To: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> References: <8d5f57670912101026r2e5b9f3ap63e13dedb497c4c@mail.gmail.com> Message-ID: <20091210223258.GA14106@hijacked.us> On Thu, Dec 10, 2009 at 10:26:39AM -0800, Shaun Clark wrote: > I have an application where I would like to route both calls and other > requests through the same queue to the same agents, for example the same > agent might take a call and then right after that take a chat. But, the chat > server we use is separate from our phone system. > > What I would like to do is basically route some text, i.e. "new chat > chat_id_goes_here" through to the agent. Is this possible with FreeSwitch? > The idea being the soft-phone would receive this text and we would write > code to catch this message do the appropriate action on our CRM. Thanks! I did this by writing my own external queueing (in erlang) and simply parking the calls in FS and adding them to my external queue (along with emails, voicemails, etc). With asterisk I added a fake call to the queue with some channel variables that referenced the external data I was really putting in the queue and I listened for the 'BRIDGE' event on the AMI and sent the agent the external data then. I'm not sure mod_fifo needs to be a universal queue - but maybe you could do what you want via api_after_bridge and uuid_chat or something crazy? You'd have to script whatever soft-phone you're using to be smart about that though. Andrew From msc at freeswitch.org Thu Dec 10 14:34:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 14:34:44 -0800 Subject: [Freeswitch-users] FS Rocks!!!!!!!!! In-Reply-To: References: <26716612.post@talk.nabble.com> <26725860.post@talk.nabble.com> <87f2f3b90912100953y7cf70387xd1714418ecef5313@mail.gmail.com> <87f2f3b90912101026j1ef7e70m7988ce6eb4c3fdb1@mail.gmail.com> <1260479480.12078.18.camel@local.freepabx.com> <87f2f3b90912101319q4b199df4y45b51611f7b019f@mail.gmail.com> Message-ID: <87f2f3b90912101434j4f4f9d9bm67a5f789b0cadd@mail.gmail.com> On Thu, Dec 10, 2009 at 2:16 PM, Julian Lyndon-Smith wrote: > Ok. The journey begins. > > http://makingfs.blogspot.com/ > > Don't know if you want to add this link to the website or wiki. > > Julian > > Excellent work! Thanks, -MC Asterisk deadlocked Why does it suck so badly? Use FreeSWITCH instead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/3c1658c8/attachment-0002.html From ken at ksac.com Thu Dec 10 16:30:03 2009 From: ken at ksac.com (Kendall Stauffer) Date: Thu, 10 Dec 2009 16:30:03 -0800 Subject: [Freeswitch-users] windows pre compiled asr Message-ID: I downloaded yesterdays latest pre compiled and seems to works great, but I get invalid Asr module when trying to run pizza app. It seemed to come pre configured with pocketsphynx, anything I should know before I spend a boat load of time on it? Rest seems real good,. thatks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/7100ed86/attachment-0002.html From ken at ksac.com Thu Dec 10 16:37:02 2009 From: ken at ksac.com (Kendall Stauffer) Date: Thu, 10 Dec 2009 16:37:02 -0800 Subject: [Freeswitch-users] never mind Message-ID: Sorry, I now see it wasn't loaded, so must not come with the pre compiled. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/71caf1a8/attachment-0002.html From msc at freeswitch.org Thu Dec 10 16:49:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Dec 2009 16:49:23 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Dec 11 Message-ID: <87f2f3b90912101649x500c0bc9kfaefc770fbf25a8@mail.gmail.com> FYI, Here's the agenda for tomorrow's conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11 Please be ready to join at 11AM CST! :) Don't forget to bring your agenda items, questions, and a willingness to help out with our various janitor projects. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/256f225a/attachment-0002.html From brian at microcomaustralia.com.au Thu Dec 10 17:57:24 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Fri, 11 Dec 2009 12:57:24 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <4B207ECC.1020405@microcomaustralia.com.au> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> Message-ID: <20091211015724.GC14547@sys11.in.vpac.org> On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: > Lack of OpenZAP support might be an issue, I assume that would be > required to connect to an onboard analogue port... I assume I could just > install Debian or another distribution instead though. This is another distribution I found: http://linux.voyage.hk/ It comes with Asterisk out of the box, although I suspect it wouldn't be too hard to get Freeswitch working instead. -- Brian May From mike at jerris.com Thu Dec 10 18:20:39 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Dec 2009 21:20:39 -0500 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211015724.GC14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: As a note, we are pretty aggressive about making sure all this stuff works right out of svn without any patches so it should be easy to port freeswitch to most platforms now. Mike On Dec 10, 2009, at 8:57 PM, Brian May wrote: > On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: >> Lack of OpenZAP support might be an issue, I assume that would be >> required to connect to an onboard analogue port... I assume I could just >> install Debian or another distribution instead though. > > This is another distribution I found: > > http://linux.voyage.hk/ > > It comes with Asterisk out of the box, although I suspect it > wouldn't be too hard to get Freeswitch working instead. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at microcomaustralia.com.au Thu Dec 10 18:42:28 2009 From: brian at microcomaustralia.com.au (Brian May) Date: Fri, 11 Dec 2009 13:42:28 +1100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: <20091211024228.GE14547@sys11.in.vpac.org> On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote: > As a note, we are pretty aggressive about making sure all this stuff works > right out of svn without any patches so it should be easy to port freeswitch > to most platforms now. Thats good to hear. I am guessing this means I should use a recent version. I see there is an Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should be able to build from the source. Anyway I sent an email to Yawarra to ask them if the net5501 computer is compatible with the TDM400 cards. There is something about a kit for the dual rack mount computer for the TDM400, which would be good if I had a rack, and somewhere to put a rack. So presumably this means it should work for the non-rack mount system too. -- Brian May From JCasale at activenetwerx.com Thu Dec 10 19:30:27 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Fri, 11 Dec 2009 03:30:27 +0000 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211024228.GE14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> <20091211024228.GE14547@sys11.in.vpac.org> Message-ID: >Anyway I sent an email to Yawarra to ask them if the net5501 computer > is >compatible with the TDM400 cards. It is, people have been doing this for a while w/ astlinux: http://www.mail-archive.com/astlinux-users at lists.sourceforge.net/msg03048.html jlc From mcampbellsmith at gmail.com Thu Dec 10 20:10:26 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 11 Dec 2009 15:10:26 +1100 Subject: [Freeswitch-users] Passing user variables to mod_voicemail Message-ID: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Hi! My voip provider provides a SOAP interface to be able to send SMS's, so after a voicemail is left, I want to execute a 'send sms' script. I don't want a separate statement in the dialplan after the voicemail statement because I only want to send sms's when a voicemail is actually left. The way I was going to do this was to modify the mailer-app to point to a shell script and modify the mailer-app-args to include some user defined variables (in conf/directory/default/*.xml). The shell script would do the following: emailvm.sh #$1 $2 $3 = smsaccount smspassword textmessage tee /tmp/vmmail | /usr/sbin/sendmail -t exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log However, if I uncomment the last line, I never see the user variables being passed to the shell script. The email is sucessfully sent, but the sms script doesnt work. If fact, the output of log.log is (for example): -f 1001 at 192.168.1.120 email_address at domain.com Any ideas if it is possible to pass user variables via mod_voicemail in this way? Thanks From anthony.minessale at gmail.com Thu Dec 10 21:28:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Dec 2009 23:28:17 -0600 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Message-ID: <191c3a030912102128n76e35268l7aa057fd1ff205d7@mail.gmail.com> That wont work. I'm not sure if there is a way, I cant think of one off the top of my head. On Thu, Dec 10, 2009 at 10:10 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > My voip provider provides a SOAP interface to be able to send SMS's, > so after a voicemail is left, I want to execute a 'send sms' script. > I don't want a separate statement in the dialplan after the voicemail > statement because I only want to send sms's when a voicemail is > actually left. > > The way I was going to do this was to modify the mailer-app to point > to a shell script and modify the mailer-app-args to include some user > defined variables (in conf/directory/default/*.xml). > > value="/usr/local/freeswitch/scripts/emailvm.sh"/> > > > The shell script would do the following: > > emailvm.sh > > #$1 $2 $3 = smsaccount smspassword textmessage > tee /tmp/vmmail | /usr/sbin/sendmail -t > exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 > #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log > > However, if I uncomment the last line, I never see the user variables > being passed to the shell script. The email is sucessfully sent, but > the sms script doesnt work. If fact, the output of log.log is (for > example): > > -f 1001 at 192.168.1.120 email_address at domain.com > > Any ideas if it is possible to pass user variables via mod_voicemail > in this way? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091210/289b8826/attachment-0002.html From codecomplete at free.fr Fri Dec 11 00:46:00 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 00:46:00 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> Message-ID: <26740589.post@talk.nabble.com> Michael Jerris wrote: > we also support natpmp and static ip setting. What is "static ip setting"? Telling FS what the public IP is? If that's what it is, what about the UDP ports that must be open to allow incoming connections? So, in the case where the FS server is located in a private network, these are the ways to open up the ports it needs to allow remote SIP users to connect to it: - UPnP and NAT-PMP (FS asks the router for its public IP, and negotiates opening the required UDP ports dynamically) - STUN (to get the public IP address from a remote STUN server) + port-mapping (to permanently open required UDP ports on NAT firewall) - possibly this fourth solution above -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26740589.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Fri Dec 11 00:59:03 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 11 Dec 2009 09:59:03 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211015724.GC14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> Message-ID: <65d96fc80912110059u5b2b7a5h389421716563aa54@mail.gmail.com> voyage linux is a stripped debian and i was using it on an alix board some time ago... Asterisk was compiling on that without any issue. I beleive FS will do the same. T. On Fri, Dec 11, 2009 at 2:57 AM, Brian May wrote: > On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: > > Lack of OpenZAP support might be an issue, I assume that would be > > required to connect to an onboard analogue port... I assume I could just > > install Debian or another distribution instead though. > > This is another distribution I found: > > http://linux.voyage.hk/ > > It comes with Asterisk out of the box, although I suspect it > wouldn't be too hard to get Freeswitch working instead. > -- > Brian May > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/969cfd3e/attachment-0002.html From mctch at yahoo.com Fri Dec 11 01:44:12 2009 From: mctch at yahoo.com (Mark Crane) Date: Fri, 11 Dec 2009 01:44:12 -0800 (PST) Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <4B213D87020000E100000322@mail.fribert.dk> Message-ID: <117186.49301.qm@web56406.mail.re3.yahoo.com> What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark ? The extensions are located under directory/default, and they look like this: ? ??? ????? ????? ????? ????? ????? ????? ??? ??? ????? ????? ????? ????? ????? ??? ? ? ??? ????? ????? ????? ????? ????? ????? ??? ??? ????? ????? ????? ??? ? As I understand them, the context set there, is the right one? ? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. ????? Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! ? It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? ? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml ? ? The default.xml looks like this ( I haven't changed it): ? ? ? ??? ????? ????? ? ????? ??? ? ??? ??? ????? ????? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ????? ? ????? ????? ????? ????? ? ? ? ????? ? ????? ? ? ? ????? ??? ? ??? ??? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ? ????? ??? ??? ??? ?? ??? ??? ????? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ????? ? ????? ??? ? ??? ? ? ??? ??? ????? ? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ? ????? ??? ??? ??? ??? ????? ? ????? ??? ? ??? ?????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ??? ??? ????? ? ? ????? ??? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ??????? ??????? ? ????? ??? ? ??? ??? ????? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ????? ? ????? ??? ??? ??? ????? ????? ????? ? ? ????? ??? ? ??? ??? ????? ????? ????? ?]]> ? ????? ??? ??? ??? ??? ????? ????? ????? ? ? ?????? ??? ? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ??? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ??? ????? ? ? ? ????? ??? ? ??? ????? ??????? ??????? ????? ??? ? ??? ????? ? ? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ????? ? ? ????? ??? ? ??? ? ??? ????? ????? ? ? ? ? ? ? ? ????? ??? ? ??? ??? ??? ??? ? ? ??? ??? ????? ??? ? ? ??? ??? ??? ? ??? ? ? ? ? Then I have under default dir: musimidk.xml ?? ?????? ?? and 9000_recordings.xml ?? ?????? ?? ? ? Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? ? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/c6baee33/attachment-0002.html From Russell.Mosemann at cune.org Fri Dec 11 04:33:45 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Fri, 11 Dec 2009 06:33:45 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26740589.post@talk.nabble.com> References: <26727762.post@talk.nabble.com><26731188.post@talk.nabble.com><0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> Message-ID: Fred-145 wrote: > What is "static ip setting"? Telling FS what the public IP is? If that's > what it is, what about the UDP ports that must be open to allow incoming > connections? Yes, static IP setting puts the (non-changing) IP addresses in the FS configuration. The ports must be manually opened/forwarded in the firewall. -- Russell Mosemann From pjintheusa at gmail.com Fri Dec 11 05:52:21 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 11 Dec 2009 08:52:21 -0500 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> Message-ID: <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> Hi - sorry to go off topic - but we are looking for Voip supplier with SMS capability. Would you mind telling me which Voip supplier you use? On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > My voip provider provides a SOAP interface to be able to send SMS's, > so after a voicemail is left, I want to execute a 'send sms' script. > I don't want a separate statement in the dialplan after the voicemail > statement because I only want to send sms's when a voicemail is > actually left. > > The way I was going to do this was to modify the mailer-app to point > to a shell script and modify the mailer-app-args to include some user > defined variables (in conf/directory/default/*.xml). > > value="/usr/local/freeswitch/scripts/emailvm.sh"/> > > > The shell script would do the following: > > emailvm.sh > > #$1 $2 $3 = smsaccount smspassword textmessage > tee /tmp/vmmail | /usr/sbin/sendmail -t > exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 > #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log > > However, if I uncomment the last line, I never see the user variables > being passed to the shell script. The email is sucessfully sent, but > the sms script doesnt work. If fact, the output of log.log is (for > example): > > -f 1001 at 192.168.1.120 email_address at domain.com > > Any ideas if it is possible to pass user variables via mod_voicemail > in this way? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/6fc32154/attachment-0002.html From mcampbellsmith at gmail.com Fri Dec 11 06:28:03 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 12 Dec 2009 01:28:03 +1100 Subject: [Freeswitch-users] Passing user variables to mod_voicemail In-Reply-To: <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> References: <33c87fa30912102010j69ee044ey8143bbc2798320ff@mail.gmail.com> <367751820912110552i6e15f4ecl7b73cfe09609ee2f@mail.gmail.com> Message-ID: <33c87fa30912110628r14923948q11909e9175f768bc@mail.gmail.com> Pennytel.com On Sat, Dec 12, 2009 at 12:52 AM, Phillip Jones wrote: > Hi - sorry to go off topic - but we are looking for Voip supplier with SMS > capability. Would you mind telling me which Voip supplier you use? > > On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> My voip provider provides a SOAP interface to be able to send SMS's, >> so after a voicemail is left, I want to execute a 'send sms' script. >> I don't want a separate statement in the dialplan after the voicemail >> statement because I only want to send sms's when a voicemail is >> actually left. >> >> The way I was going to do this was to modify the mailer-app to point >> to a shell script and modify the mailer-app-args to include some user >> defined variables (in conf/directory/default/*.xml). >> >> ? ?> value="/usr/local/freeswitch/scripts/emailvm.sh"/> >> ? ? >> >> The shell script would do the following: >> >> emailvm.sh >> >> #$1 $2 $3 = smsaccount smspassword textmessage >> tee /tmp/vmmail | /usr/sbin/sendmail -t >> exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3 >> #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log >> >> However, if I uncomment the last line, I never see the user variables >> being passed to the shell script. ?The email is sucessfully sent, but >> the sms script doesnt work. ?If fact, the output of log.log is (for >> example): >> >> -f 1001 at 192.168.1.120 email_address at domain.com >> >> Any ideas if it is possible to pass user variables via mod_voicemail >> in this way? >> >> Thanks >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ken at ksac.com Fri Dec 11 07:47:55 2009 From: ken at ksac.com (Kendall Stauffer) Date: Fri, 11 Dec 2009 07:47:55 -0800 Subject: [Freeswitch-users] Still cant find it Message-ID: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/021854da/attachment-0002.html From chris.chen2004 at gmail.com Fri Dec 11 08:06:09 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 11:06:09 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks Message-ID: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> Hi there, I have very strange behaviors for my SIP endpoints with FS SVN trunk 15905. I tested with both Polycom IP650 and Bria 2.5.4, compared against port audio and googletalk endpoints in the same network. all SIP end points (Polycom and Bria) behind NAT but in the same subnet 192.168.0, I tried to change the settings below: in /conf/sip_profiles/internal.xml using different combinations of either enabling or disabling them. the results are all the same, the audios on sip endpoints always got cut about 31 seconds, no issues at all with either port audio or gtalk, Could anyone point me to the right direction for the sofia_sip profile setup? Your helps are greatly appreciated Thanks, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/76e54d74/attachment-0002.html From brian at freeswitch.org Fri Dec 11 08:08:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 10:08:47 -0600 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: > Ok, So I have looked around a lot now, think I have read everything > carefully, and don?t see an answer to my questions anywhere, but > apologize if it is already somewhere. > SO I need the sphinx and tts modules, and don?t see their src on > the site with the freeswitch stuff. > Do I just download from CMU? Any certain versions? > Would be nice if somebody already compiled for windows > I am very impressed with freeswitch, and thank you for your > efforts!!! > > _______________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/91a1d3bc/attachment-0002.html From jeff at jefflenk.com Fri Dec 11 08:23:05 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 11 Dec 2009 08:23:05 -0800 (PST) Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: <1260548585714-4152195.post@n2.nabble.com> The source tarballs are downloaded by the vs2008 project files when you build the solution Kendall Stauffer wrote: > > Ok, So I have looked around a lot now, think I have read everything > carefully, and don't see an answer to my questions anywhere, but apologize > if it is already somewhere. > SO I need the sphinx and tts modules, and don't see their src on the > site with the freeswitch stuff. > Do I just download from CMU? Any certain versions? > Would be nice if somebody already compiled for windows > I am very impressed with freeswitch, and thank you for your efforts!!! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Still-cant-find-it-tp4152031p4152195.html Sent from the freeswitch-users mailing list archive at Nabble.com. From frank at carmickle.com Fri Dec 11 08:25:00 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 11 Dec 2009 11:25:00 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> Message-ID: <20091211162459.GR31924@base.carmickle.com> On Fri, Dec 11, Chris Chen wrote: > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > trunk 15905. Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. --FC From ken at ksac.com Fri Dec 11 08:25:33 2009 From: ken at ksac.com (Kendall Stauffer) Date: Fri, 11 Dec 2009 08:25:33 -0800 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: Yes, I can do that , I don't see where I download the source, Sorry to bug you. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, December 11, 2009 11:09 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Still cant find it Download MSVC and compile it yourself is usually the best bet. /b On Dec 11, 2009, at 9:47 AM, Kendall Stauffer wrote: Ok, So I have looked around a lot now, think I have read everything carefully, and don't see an answer to my questions anywhere, but apologize if it is already somewhere. SO I need the sphinx and tts modules, and don't see their src on the site with the freeswitch stuff. Do I just download from CMU? Any certain versions? Would be nice if somebody already compiled for windows I am very impressed with freeswitch, and thank you for your efforts!!! _______________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/42877b15/attachment-0002.html From chris.chen2004 at gmail.com Fri Dec 11 08:44:27 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 11:44:27 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <20091211162459.GR31924@base.carmickle.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> Message-ID: <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly I noticed that the acl automatically having 192.168.0.0 set as "deny", that's why I tried to changed the settings regarding nat acl and localnet acl. Chris On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > On Fri, Dec 11, Chris Chen wrote: > > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > > trunk 15905. > > Is this a change in behavior or is this the first time you've run > freeswitch? If this is your first time welcome aboard! Also if this is > your first time you've probably have some IPs aliased on your interface and > you still have stun enabled. This was the behavior I saw the first time I > ran it on a box with aliases on an interface. The stun server tells > freeswitch after some time that the IP is different then the one you've > assigned. This is just one possibility. If this isn't the case then we > will need to see sip traces on all of your profiles. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/89ffe944/attachment-0002.html From carlos.talbot at gmail.com Fri Dec 11 09:04:06 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Fri, 11 Dec 2009 11:04:06 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: References: Message-ID: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> It hasn't been included as of late since I'm getting an unresolved link error during the build. I'll need someone experienced in pocketsphinx to assist with this issue: 13>ngram_search.obj : error LNK2001: unresolved external symbol _ngram_model_flush 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: 1 unresolved externals regards, Carlos On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer wrote: > I downloaded yesterdays latest pre compiled and seems to works great, but > I get invalid Asr module when trying to run pizza app. > > It seemed to come pre configured with pocketsphynx, anything I should > know before I spend a boat load of time on it? > > Rest seems real good,. thatks!!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/23e8e033/attachment-0002.html From msc at freeswitch.org Fri Dec 11 09:04:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Dec 2009 09:04:07 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting! Message-ID: <87f2f3b90912110904y716e85f9o6afa42a1960da70f@mail.gmail.com> Come one, come all! http://bit.ly/8KzHCZ Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/75833c93/attachment-0002.html From brian at freeswitch.org Fri Dec 11 09:08:10 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 11:08:10 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> References: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> Message-ID: <28E7DE31-320E-47D7-86CD-40619CF37269@freeswitch.org> Thats being fixed today! ;) /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: > It hasn't been included as of late since I'm getting an unresolved > link error during the build. I'll need someone experienced in > pocketsphinx to assist with this issue: > > 13>ngram_search.obj : error LNK2001: unresolved external symbol > _ngram_model_flush > 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: > 1 unresolved externals > > regards, > > Carlos > From kristian.kielhofner at gmail.com Fri Dec 11 09:14:25 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 11 Dec 2009 12:14:25 -0500 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket Message-ID: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> Hello everyone, PB here: http://pastebin.freeswitch.org/11482 FS rev 15909. The relevant bits from the log are here (starting around line 135): # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 sofia/pjsip/nobody at 192.168.4.253 Command Execute bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj at 192.168.4.17) # EXECUTE sofia/pjsip/nobody at 192.168.4.253 bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara Ousley}sofia/voalte/huttoj at 192.168.4.17) # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 0 = [originate_timeout=30] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 1 = [bypass_media=true] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 2 = [origination_caller_id_number=1111] # 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 variable string 3 = [origination_caller_id_name=Tara Ousley] bypass_media=true yet the SDP of the outgoing INVITE looks like this: # send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: # ------------------------------------------------------------------------ # INVITE sip:huttoj at 192.168.4.17 SIP/2.0 # Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr # Max-Forwards: 69 # From: "Tara Ousley" ;tag=0tU8SN9pvejNK # To: # Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 # CSeq: 124148250 INVITE # Contact: # User-Agent: Voalte Voice # Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY # Supported: timer, precondition, path, replaces # Allow-Events: talk, refer # Content-Type: application/sdp # Content-Disposition: session # Content-Length: 271 # X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 # Remote-Party-ID: "Tara Ousley" ;party=calling;screen=yes;privacy=off # # v=0 # o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 # s=FreeSWITCH # c=IN IP4 192.168.2.10 # t=0 0 # m=audio 25172 RTP/AVP 9 0 101 # a=rtpmap:9 G722/8000 # a=rtpmap:0 PCMU/8000 # a=rtpmap:101 telephone-event/8000 # a=fmtp:101 0-16 # a=silenceSupp:off - - - - # a=ptime:20 192.168.2.10 is the address of my FS box... -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Fri Dec 11 09:31:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Dec 2009 11:31:26 -0600 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket In-Reply-To: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> References: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> Message-ID: <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> Hey, You can't set bypass_media=true in {} or it will not take effect unless that b leg itself becomes an a leg some day. you need to execute set on bypass_media=true on the leg before you call bridge to trigger it. Alternatively you could set {bypass_media_after_bridge=true} or set it on A leg as described above on either leg and it will do the bypass once the audio is flowing. On Fri, Dec 11, 2009 at 11:14 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > PB here: > > http://pastebin.freeswitch.org/11482 > > FS rev 15909. The relevant bits from the log are here (starting > around line 135): > > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr.c:548 > sofia/pjsip/nobody at 192.168.4.253 Command Execute > > bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara > Ousley}sofia/voalte/huttoj at 192.168.4.17) > # > EXECUTE sofia/pjsip/nobody at 192.168.4.253 > > bridge({originate_timeout=30,bypass_media=true,origination_caller_id_number=1111,origination_caller_id_name=Tara > Ousley}sofia/voalte/huttoj at 192.168.4.17) > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 0 = [originate_timeout=30] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 1 = [bypass_media=true] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 2 = [origination_caller_id_number=1111] > # > 2009-12-11 12:06:12.875158 [DEBUG] switch_ivr_originate.c:1735 > variable string 3 = [origination_caller_id_name=Tara Ousley] > > bypass_media=true yet the SDP of the outgoing INVITE looks like this: > > # > send 1032 bytes to udp/[192.168.4.17]:5060 at 12:06:12.876994: > # > ------------------------------------------------------------------------ > # > INVITE sip:huttoj at 192.168.4.17 SIP/2.0 > # > Via: SIP/2.0/UDP 192.168.2.10:5062;rport;branch=z9hG4bK1K6mc3NcmmaNr > # > Max-Forwards: 69 > # > From: "Tara Ousley" > >;tag=0tU8SN9pvejNK > # > To: > > # > Call-ID: 802f4045-4215-42a2-91a6-ff9cf18b1aa8 > # > CSeq: 124148250 INVITE > # > Contact: > # > User-Agent: Voalte Voice > # > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > # > Supported: timer, precondition, path, replaces > # > Allow-Events: talk, refer > # > Content-Type: application/sdp > # > Content-Disposition: session > # > Content-Length: 271 > # > X-voalte-call-id: 898ef33c-50f2-487c-9e8d-8c6fcee15ab8 > # > Remote-Party-ID: "Tara Ousley" > > >;party=calling;screen=yes;privacy=off > # > > # > v=0 > # > o=FreeSWITCH 1260508000 1260508001 IN IP4 192.168.2.10 > # > s=FreeSWITCH > # > c=IN IP4 192.168.2.10 > # > t=0 0 > # > m=audio 25172 RTP/AVP 9 0 101 > # > a=rtpmap:9 G722/8000 > # > a=rtpmap:0 PCMU/8000 > # > a=rtpmap:101 telephone-event/8000 > # > a=fmtp:101 0-16 > # > a=silenceSupp:off - - - - > # > a=ptime:20 > > 192.168.2.10 is the address of my FS box... > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/35e93a9f/attachment-0002.html From zendel.fernandez at gmail.com Fri Dec 11 03:02:47 2009 From: zendel.fernandez at gmail.com (zendel fernandez) Date: Fri, 11 Dec 2009 16:32:47 +0530 Subject: [Freeswitch-users] gtalk dingaling G723 Message-ID: hi! Pls shed some light to the below dingaling/gtalk issue. ___________________________Call path________________________ Gtalk ---> Account setup in client.xml ---routed to public.xml ---> route to a external SIP gateway -----> sipUserA ___________________________Problem statement________________ The above call scenario is successfull using the *PMCU, PCMA* codecs in "dingaling.conf.xml". Both parties can hear each other. When I choose G723 the call disconnects after few seconds(2~). Both parties can hear some noise & thasts all. I see the call is bridged and then immediately unbridged. I assume G.723 is suppose to work here in passtru mode. Do I have to do any other configuration ohter than the ones listed below in order to get passthru working. ___________________________LOG important parts________________ (Full log found at the end) 2009-12-11 16:05:12.948700 [ERR] mod_g723_1.c:148 This codec is only usable in passthrough mode! 2009-12-11 16:05:12.948700 [DEBUG] switch_ivr_bridge.c:464 DingaLing/new ending bridge by request from write function 2009-12-11 16:05:12.948700 [DEBUG] switch_ivr_bridge.c:520 sofia/external/0094777915380 receive message [UNBRIDGE] ________________________Configuration XMLS______________________ ________________________dingaling.conf.xml_____________________ ______________________________public.xml__________________________ ______________________________client.xml___________________________ __________________________________Full Log__________________________ ____________________________________________________________________ ____________________________________________________________________ ____________________________________________________________________ freeswitch at internal> 2009-12-11 16:18:05.848176 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:353 Created Session 596627090 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [ISAC] id='103' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [IPCMWB] id='97' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [G723] id='4' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [EG711U] id='100' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [EG711A] id='101' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [PCMU] id='0' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [PCMA] id='8' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [CN] id='13' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [iLBC] id='102' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [red] id='117' 2009-12-11 16:18:05.848176 [DEBUG] libdingaling.c:421 Add Payload [audio/telephone-event] id='106' 2009-12-11 16:18:05.848176 [DEBUG] mod_dingaling.c:3028 Creating an identity for 596627090 GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB < GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB> 5555 2009-12-11 16:18:05.848176 [NOTICE] switch_channel.c:613 New Channel dingaling/5555 [d689728e-fcbd-433d-8b87-7c54c03d2825] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3056 Creating a session for 596627090 2009-12-11 16:18:05.849387 [NOTICE] switch_channel.c:611 Rename Channel dingaling/5555->DingaLing/new [d689728e-fcbd-433d-8b87-7c54c03d2825] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3060 (DingaLing/new) State Change CS_NEW -> CS_INIT 2009-12-11 16:18:05.849387 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3157 11 payloads 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload ISAC 103 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare ISAC 103/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload IPCMWB 97 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare IPCMWB 97/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3159 Available Payload G723 4 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3166 compare G723 4/8000 to G723 4/8000 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:3177 Choosing Payload index 0 G723 4 2009-12-11 16:18:05.849387 [DEBUG] mod_dingaling.c:1066 Send Describe [G723 at 8000] 2009-12-11 16:18:05.849387 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_INIT 2009-12-11 16:18:05.849387 [DEBUG] switch_core_state_machine.c:338 (DingaLing/new) State INIT 2009-12-11 16:18:05.849387 [NOTICE] mod_dingaling.c:1093 Ring-Ready DingaLing/new! 2009-12-11 16:18:05.884166 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.884166 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:05.884166 [DEBUG] libdingaling.c:1406 Sending packet 300 (2 left) 2009-12-11 16:18:05.884166 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:05.983333 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.084116 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.184092 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.284068 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.384042 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.483334 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.583993 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.683965 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.783940 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.867919 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:06.867919 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:06.867919 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=Unokk7lKh84I8mEM password=At5OQzApw/ZrIrCo address=172.16.11.110 port=1563 pref=1.00 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:3227 1 candidates 2009-12-11 16:18:06.867919 [DEBUG] mod_dingaling.c:3243 candidate 172.16.11.110:1563 FAIL ACL wan 2009-12-11 16:18:06.883915 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:06.883915 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:06.983330 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.013899 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:07.013899 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=W3xgvc25vCXb5M/U password=7EcqsVB3S6J3I2+g address=172.16.7.28 port=1566 pref=1.00 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:3227 1 candidates 2009-12-11 16:18:07.013899 [DEBUG] mod_dingaling.c:3243 candidate 172.16.7.28:1566 FAIL ACL wan 2009-12-11 16:18:07.083870 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.083870 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.119862 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:07.119862 [DEBUG] libdingaling.c:943 Cancel packet 300 2009-12-11 16:18:07.183848 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:07.183848 [DEBUG] libdingaling.c:1414 Discarding packet 300 2009-12-11 16:18:08.365552 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:08.365552 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:08.365552 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=relay protocol=udp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=19295 pref=0.50 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3227 2 candidates 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3243 candidate 172.16.7.28:1566 FAIL ACL wan 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3239 candidate 209.85.229.126:19295 PASS ACL wan 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:3288 Acceptable Candidate 209.85.229.126:19295 2009-12-11 16:18:08.365552 [DEBUG] mod_dingaling.c:976 Stun Lookup Local 172.16.11.211:22878 2009-12-11 16:18:15.496911 [INFO] mod_dingaling.c:984 Stun Success XX.XX.XX.XX:22878 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:998 Send Candidate XX.XX.XX.XX:22878 [goTRR95uTkRRMe3t] 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:856 Set Read Codec to G723 at 8000 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:871 Set Write Codec to G723 at 8000 2009-12-11 16:18:15.496911 [DEBUG] mod_dingaling.c:884 SETUP RTP 172.16.11.211:22878 -> 209.85.229.126:19295 2009-12-11 16:18:15.496911 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 960 bytes per 120ms 2009-12-11 16:18:15.498911 [DEBUG] switch_rtp.c:2905 Activate VAD codec G723 120ms 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1184 (DingaLing/new) State Change CS_INIT -> CS_ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:338 (DingaLing/new) State INIT going to sleep 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:341 (DingaLing/new) State ROUTING 2009-12-11 16:18:15.498911 [DEBUG] mod_dingaling.c:1198 DingaLing/new CHANNEL ROUTING 2009-12-11 16:18:15.498911 [DEBUG] switch_core_state_machine.c:78 DingaLing/new Standard ROUTING 2009-12-11 16:18:15.498911 [INFO] mod_dialplan_xml.c:408 Processing GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB->5555 in context public Dialplan: DingaLing/new parsing [public->public_did] continue=false Dialplan: DingaLing/new Regex (PASS) [public_did] caller_id_number( GTALK_ANY_CLIENT at gmail.com/Talk.v1054D5EA6CB) =~ /^([^@]+)/ break=never Dialplan: DingaLing/new Action set(effective_caller_id_number=GTALK_ANY_CLIENT) Dialplan: DingaLing/new Regex (PASS) [public_did] destination_number(5555) =~ /^(5555)$/ break=on-false Dialplan: DingaLing/new Action set(call_timeout=18) Dialplan: DingaLing/new Action set(continue_on_fail=true) Dialplan: DingaLing/new Action set(hangup_after_bridge=true) Dialplan: DingaLing/new Action bridge(sofia/gateway/sbc/sipUserA) Dialplan: DingaLing/new Action answer() Dialplan: DingaLing/new Action voicemail(default 172.16.11.211 1001) 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:122 (DingaLing/new) State Change CS_ROUTING -> CS_EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:15.499911 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:341 (DingaLing/new) State ROUTING going to sleep 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:348 (DingaLing/new) State EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] mod_dingaling.c:1215 DingaLing/new CHANNEL EXECUTE 2009-12-11 16:18:15.499911 [DEBUG] switch_core_state_machine.c:159 DingaLing/new Standard EXECUTE EXECUTE DingaLing/new set(effective_caller_id_number=GTALK_ANY_CLIENT) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [effective_caller_id_number]=[GTALK_ANY_CLIENT] EXECUTE DingaLing/new set(call_timeout=18) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [call_timeout]=[18] EXECUTE DingaLing/new set(continue_on_fail=true) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [continue_on_fail]=[true] EXECUTE DingaLing/new set(hangup_after_bridge=true) 2009-12-11 16:18:15.499911 [DEBUG] mod_dptools.c:768 DingaLing/new SET [hangup_after_bridge]=[true] EXECUTE DingaLing/new bridge(sofia/gateway/sbc/sipUserA) 2009-12-11 16:18:15.500912 [NOTICE] switch_channel.c:613 New Channel sofia/external/sipUserA [cf250379-b2c9-439e-8295-766feef6cf74] 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:3142 (sofia/external/sipUserA) State Change CS_NEW -> CS_INIT 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_INIT 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:338 (sofia/external/sipUserA) State INIT 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:83 sofia/external/sipUserA SOFIA INIT 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:111 (sofia/external/sipUserA) State Change CS_INIT -> CS_ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:338 (sofia/external/sipUserA) State INIT going to sleep 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_core_state_machine.c:341 (sofia/external/sipUserA) State ROUTING 2009-12-11 16:18:15.500912 [DEBUG] mod_sofia.c:132 sofia/external/sipUserA SOFIA ROUTING 2009-12-11 16:18:15.500912 [DEBUG] switch_ivr_originate.c:66 (sofia/external/sipUserA) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-12-11 16:18:15.500912 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:341 (sofia/external/sipUserA) State ROUTING going to sleep 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_CONSUME_MEDIA 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:360 (sofia/external/sipUserA) State CONSUME_MEDIA 2009-12-11 16:18:15.501911 [DEBUG] switch_core_state_machine.c:360 (sofia/external/sipUserA) State CONSUME_MEDIA going to sleep 2009-12-11 16:18:15.501911 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [calling][0] 2009-12-11 16:18:15.583888 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:15.583888 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.583888 [DEBUG] libdingaling.c:1406 Sending packet 301 (2 left) 2009-12-11 16:18:15.583888 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:15.684864 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.784839 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.884815 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:15.984789 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.084764 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.184740 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.284715 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.384691 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.484666 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.584640 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.684617 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.784591 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.810584 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:16.810584 [DEBUG] libdingaling.c:943 Cancel packet 301 2009-12-11 16:18:16.884567 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:16.884567 [DEBUG] libdingaling.c:1414 Discarding packet 301 2009-12-11 16:18:17.795342 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.795342 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:17.795342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=local protocol=tcp username=pBOc+m0nq7uQngnq password=HzZY2+zb/u9oRH++ address=172.16.7.28 port=1569 pref=0.80 2009-12-11 16:18:17.795342 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:17.795342 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:17.798342 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=tcp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=19294 pref=0.50 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:17.798342 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=local protocol=tcp username=EXe2XpIc76S9WxgB password=BKF7SUs/hc30Fplp address=172.16.11.110 port=1570 pref=0.80 2009-12-11 16:18:17.798342 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:17.798342 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:17.884322 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:17.884322 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:18.797094 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:463 Duplicate Pref! 2009-12-11 16:18:18.797094 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=ssltcp username=erYq6ggXdBRbEVDe password=GQcop4clv7IEVldS address=209.85.229.126 port=443 pref=0.50 2009-12-11 16:18:18.797094 [DEBUG] mod_dingaling.c:2916 using Existing session for 596627090 2009-12-11 16:18:18.797094 [DEBUG] mod_dingaling.c:3223 Already picked an IP [209.85.229.126] 2009-12-11 16:18:18.885072 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:21.529273 [INFO] sofia.c:506 Update Callee ID to "sipUserA" 2009-12-11 16:18:21.529273 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [proceeding][180] 2009-12-11 16:18:21.529273 [DEBUG] sofia.c:3738 Remote SDP: v=0 o=- 298002432 1260522569 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 15002 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=ptime:30 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[G723:4:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[PCMU:0:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[PCMA:8:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3305 Audio Codec Compare [G723:4:8000:30]/[GSM:3:8000:120] 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:3353 Substituting codec G723 at 30i@8000h 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:2143 Set Codec sofia/external/sipUserA G723/8000 30 ms 240 samples 2009-12-11 16:18:21.529273 [DEBUG] sofia_glue.c:2381 AUDIO RTP [sofia/external/sipUserA] 172.16.11.211 port 32408 -> YY.YY.YY.YY port 15002 codec: 4 ms: 30 2009-12-11 16:18:21.530280 [DEBUG] switch_rtp.c:1167 Starting timer [soft] 240 bytes per 30ms 2009-12-11 16:18:21.531273 [NOTICE] sofia_glue.c:2909 Pre-Answer sofia/external/sipUserA! 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:2020 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_originate.c:2886 DingaLing/new receive message [PROGRESS] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [NOTICE] switch_ivr_originate.c:2886 Pre-Answer DingaLing/new! 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_originate.c:2929 Originate Resulted in Success: [sofia/external/sipUserA] 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:182 sofia/external/sipUserA receive message [AUDIO_SYNC] 2009-12-11 16:18:21.531273 [DEBUG] switch_channel.c:182 DingaLing/new receive message [AUDIO_SYNC] 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1032 sofia/external/sipUserA receive message [BRIDGE] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1039 DingaLing/new receive message [BRIDGE] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:21.531273 [DEBUG] switch_ivr_bridge.c:1083 (sofia/external/sipUserA) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:21.531273 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] switch_core_state_machine.c:351 (sofia/external/sipUserA) State EXCHANGE_MEDIA 2009-12-11 16:18:21.531273 [DEBUG] mod_sofia.c:455 SOFIA LOOPBACK 2009-12-11 16:18:21.548268 [DEBUG] switch_rtp.c:2004 Correct ip/port confirmed. 2009-12-11 16:18:23.513922 [INFO] sofia.c:506 Update Callee ID to "sipUserA" 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [completing][200] 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3735 Duplicate SDP v=0 o=- 298002432 1260522569 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 15002 RTP/AVP 4 8 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=ptime:30 2009-12-11 16:18:23.513922 [DEBUG] sofia.c:3727 Channel sofia/external/sipUserA entering state [ready][200] 2009-12-11 16:18:23.513922 [DEBUG] switch_channel.c:2133 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.513922 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.513922 [NOTICE] sofia.c:4219 Channel [sofia/external/sipUserA] has been answered 2009-12-11 16:18:23.513922 [DEBUG] switch_channel.c:182 sofia/external/sipUserA receive message [AUDIO_SYNC] 2009-12-11 16:18:23.587904 [DEBUG] switch_ivr_bridge.c:394 DingaLing/new receive message [ANSWER] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.587904 [NOTICE] switch_ivr_bridge.c:394 Channel [DingaLing/new] has been answered 2009-12-11 16:18:23.587904 [DEBUG] switch_channel.c:182 DingaLing/new receive message [AUDIO_SYNC] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:706 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] switch_core_session.c:706 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:23.587904 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:23.617898 [DEBUG] switch_ivr_bridge.c:131 sofia/external/sipUserA receive message [DISPLAY] 2009-12-11 16:18:23.708875 [DEBUG] switch_ivr_bridge.c:131 DingaLing/new receive message [DISPLAY] 2009-12-11 16:18:25.568413 [DEBUG] switch_rtp.c:2004 Correct ip/port confirmed. 2009-12-11 16:18:25.568413 [ERR] mod_g723_1.c:148 This codec is only usable in passthrough mode! 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:464 DingaLing/new ending bridge by request from write function 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:520 sofia/external/sipUserA receive message [UNBRIDGE] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_session.c:645 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD DONE [sofia/external/sipUserA] 2009-12-11 16:18:25.568413 [DEBUG] switch_ivr_bridge.c:565 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.568413 [NOTICE] switch_ivr_bridge.c:617 Hangup sofia/external/sipUserA [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 16:18:25.568413 [DEBUG] switch_channel.c:1912 Send signal sofia/external/sipUserA [KILL] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:488 (sofia/external/sipUserA) State HANGUP 2009-12-11 16:18:25.568413 [DEBUG] mod_sofia.c:358 Channel sofia/external/sipUserA hanging up, cause: NORMAL_CLEARING 2009-12-11 16:18:25.568413 [DEBUG] mod_sofia.c:400 Sending BYE to sofia/external/sipUserA 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:46 sofia/external/sipUserA Standard HANGUP, cause: NORMAL_CLEARING 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:488 (sofia/external/sipUserA) State HANGUP going to sleep 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:351 (sofia/external/sipUserA) State EXCHANGE_MEDIA going to sleep 2009-12-11 16:18:25.568413 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_HANGUP 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:465 sofia/external/sipUserA handler already called, skipping state handler. 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:333 (sofia/external/sipUserA) State Change CS_HANGUP -> CS_REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:314 (sofia/external/sipUserA) Running State Change CS_REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:579 (sofia/external/sipUserA) State REPORTING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:53 sofia/external/sipUserA Standard REPORTING, cause: NORMAL_CLEARING 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:579 (sofia/external/sipUserA) State REPORTING going to sleep 2009-12-11 16:18:25.569414 [DEBUG] switch_core_state_machine.c:327 (sofia/external/sipUserA) State Change CS_REPORTING -> CS_DESTROY 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:999 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.569414 [DEBUG] switch_core_session.c:1136 Session 2 (sofia/external/sipUserA) Locked, Waiting on external entities 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:520 DingaLing/new receive message [UNBRIDGE] 2009-12-11 16:18:25.628397 [DEBUG] switch_core_session.c:645 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:563 BRIDGE THREAD DONE [DingaLing/new] 2009-12-11 16:18:25.628397 [DEBUG] switch_ivr_bridge.c:565 Send signal sofia/external/sipUserA [BREAK] 2009-12-11 16:18:25.628397 [NOTICE] switch_ivr_bridge.c:1179 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 16:18:25.628397 [DEBUG] switch_channel.c:1912 Send signal DingaLing/new [KILL] 2009-12-11 16:18:25.628397 [DEBUG] libdingaling.c:298 Destroyed Session 596627090 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:488 (DingaLing/new) State HANGUP 2009-12-11 16:18:25.628397 [DEBUG] mod_dingaling.c:1293 DingaLing/new CHANNEL HANGUP 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:46 DingaLing/new Standard HANGUP, cause: NORMAL_CLEARING 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:488 (DingaLing/new) State HANGUP going to sleep 2009-12-11 16:18:25.628397 [NOTICE] switch_core_session.c:1154 Session 2 (sofia/external/sipUserA) Ended 2009-12-11 16:18:25.628397 [NOTICE] switch_core_session.c:1156 Close Channel sofia/external/sipUserA [CS_DESTROY] 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:348 (DingaLing/new) State EXECUTE going to sleep 2009-12-11 16:18:25.628397 [DEBUG] switch_core_state_machine.c:423 (sofia/external/sipUserA) Running State Change CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_HANGUP 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipUserA) State DESTROY 2009-12-11 16:18:25.629398 [DEBUG] mod_sofia.c:293 sofia/external/sipUserA SOFIA DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:465 DingaLing/new handler already called, skipping state handler. 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:333 (DingaLing/new) State Change CS_HANGUP -> CS_REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:314 (DingaLing/new) Running State Change CS_REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipUserA Standard DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:579 (DingaLing/new) State REPORTING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (sofia/external/sipUserA) State DESTROY going to sleep 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:53 DingaLing/new Standard REPORTING, cause: NORMAL_CLEARING 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:579 (DingaLing/new) State REPORTING going to sleep 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:327 (DingaLing/new) State Change CS_REPORTING -> CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:999 Send signal DingaLing/new [BREAK] 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1333 DingaLing/new CHANNEL KILL 2009-12-11 16:18:25.629398 [DEBUG] switch_core_session.c:1136 Session 1 (DingaLing/new) Locked, Waiting on external entities 2009-12-11 16:18:25.629398 [NOTICE] switch_core_session.c:1154 Session 1 (DingaLing/new) Ended 2009-12-11 16:18:25.629398 [NOTICE] switch_core_session.c:1156 Close Channel DingaLing/new [CS_DESTROY] 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:423 (DingaLing/new) Running State Change CS_DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (DingaLing/new) State DESTROY 2009-12-11 16:18:25.629398 [DEBUG] mod_dingaling.c:1231 NUKE RTP 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:60 DingaLing/new Standard DESTROY 2009-12-11 16:18:25.629398 [DEBUG] switch_core_state_machine.c:434 (DingaLing/new) State DESTROY going to sleep 2009-12-11 16:18:25.683383 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.683383 [DEBUG] libdingaling.c:1406 Sending packet 302 (2 left) 2009-12-11 16:18:25.683383 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:25.783359 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.883332 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:25.983307 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.083280 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.183267 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.283231 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.383206 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.483181 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.583156 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.651139 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.651139 [DEBUG] libdingaling.c:943 Cancel packet 302 2009-12-11 16:18:26.660136 [INFO] libdingaling.c:1305 SecRECV: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.660136 [DEBUG] libdingaling.c:353 Created Session 596627090 2009-12-11 16:18:26.660136 [DEBUG] libdingaling.c:381 Message for Session 596627090 2009-12-11 16:18:26.660136 [DEBUG] mod_dingaling.c:2926 Session is already dead 2009-12-11 16:18:26.683132 [NOTICE] libdingaling.c:1307 SecSEND: ------------------------------------------------------------------------------- 2009-12-11 16:18:26.683132 [DEBUG] libdingaling.c:1389 Processing 1 packets in retry queue 2009-12-11 16:18:26.683132 [DEBUG] libdingaling.c:1414 Discarding packet 302 Regds. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/ac7d3bc6/attachment-0002.html From mike at jerris.com Fri Dec 11 09:51:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 12:51:13 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> Message-ID: <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> As i said multiple times on irc last night, we need to see debug logs with sip trace to see what is going on. Mike On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > Thanks Frank for sharing your experience. This is the behavior change just starting within three days, maybe because of some code changes in mod_sofia which I should change the settings accordingly > I noticed that the acl automatically having 192.168.0.0 set as "deny", that's why I tried to changed the settings regarding nat acl and localnet acl. > > Chris > > > > On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > On Fri, Dec 11, Chris Chen wrote: > > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN > > trunk 15905. > > Is this a change in behavior or is this the first time you've run freeswitch? If this is your first time welcome aboard! Also if this is your first time you've probably have some IPs aliased on your interface and you still have stun enabled. This was the behavior I saw the first time I ran it on a box with aliases on an interface. The stun server tells freeswitch after some time that the IP is different then the one you've assigned. This is just one possibility. If this isn't the case then we will need to see sip traces on all of your profiles. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/917eb7b4/attachment-0002.html From mike at jerris.com Fri Dec 11 09:55:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 12:55:13 -0500 Subject: [Freeswitch-users] Still cant find it In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code On Dec 11, 2009, at 11:25 AM, Kendall Stauffer wrote: > Yes, I can do that , I don?t see where I download the source, Sorry to bug you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/694aec32/attachment-0002.html From brian at freeswitch.org Fri Dec 11 09:56:52 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 11:56:52 -0600 Subject: [Freeswitch-users] windows pre compiled asr In-Reply-To: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> References: <5800526b0912110904j68af3e00sdb5ccbca52c9905b@mail.gmail.com> Message-ID: Can you confirm its fixed now? /b On Dec 11, 2009, at 11:04 AM, Carlos Talbot wrote: > It hasn't been included as of late since I'm getting an unresolved > link error during the build. I'll need someone experienced in > pocketsphinx to assist with this issue: > > 13>ngram_search.obj : error LNK2001: unresolved external symbol > _ngram_model_flush > 13>G:\freeswitch_dev\Release\pocketsphinx.dll : fatal error LNK1120: > 1 unresolved externals > > regards, > > Carlos > > On Thu, Dec 10, 2009 at 6:30 PM, Kendall Stauffer > wrote: > I downloaded yesterdays latest pre compiled and seems to works > great, but I get invalid Asr module when trying to run pizza app. > > It seemed to come pre configured with pocketsphynx, anything I > should know before I spend a boat load of time on it? > > Rest seems real good,. thatks!!! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/2538794a/attachment-0002.html From brian at freeswitch.org Fri Dec 11 10:03:51 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:03:51 -0600 Subject: [Freeswitch-users] gtalk dingaling G723 In-Reply-To: References: Message-ID: <958069D8-A221-4F8A-A155-430AFB141393@freeswitch.org> Can't use G723. /b On Dec 11, 2009, at 5:02 AM, zendel fernandez wrote: > > hi! > > Pls shed some light to the below dingaling/gtalk issue. > From codecomplete at free.fr Fri Dec 11 10:09:09 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 10:09:09 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> Message-ID: <26748901.post@talk.nabble.com> One last question: Does someone know of a utility for Windows that can check that a NAT router supports either UPnP or NAT-PMP? I guess it's no big deal to write a small diagnostic by connecting to free firewall checkers to see if the relevant ports are open, but if it's already available... Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26748901.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Dec 11 10:13:21 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:13:21 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26748901.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> Message-ID: <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> FreeSWITCH on windows will already poke holes in the windows firewall using upnp. Just start FS and it works. Your outer nat is a larger issue... /b On Dec 11, 2009, at 12:09 PM, Fred-145 wrote: > > One last question: Does someone know of a utility for Windows that > can check > that a NAT router supports either UPnP or NAT-PMP? I guess it's no > big deal > to write a small diagnostic by connecting to free firewall checkers > to see > if the relevant ports are open, but if it's already available... > > Thank you. > -- From asterisk at dotr.com Fri Dec 11 10:19:39 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Fri, 11 Dec 2009 18:19:39 +0000 Subject: [Freeswitch-users] The Building Freeswitch blog Message-ID: Doing the building thing, seem to have come across a bug. Have a look at Part 2 of http://makingfs.blogspot.com/ If make crashes out, it states that it was successfully built ;) Julian From brian at freeswitch.org Fri Dec 11 10:24:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 12:24:38 -0600 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: <4A6BCF50-D8E1-4D8B-BED0-6616A5BAB8A2@freeswitch.org> well mod_alas.c is for the N800 Please open a jira. /b On Dec 11, 2009, at 12:19 PM, Julian Lyndon-Smith wrote: > Doing the building thing, seem to have come across a bug. > > Have a look at Part 2 of http://makingfs.blogspot.com/ > > If make crashes out, it states that it was successfully built ;) > > Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/52960711/attachment-0002.html From mike at jerris.com Fri Dec 11 10:34:45 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 13:34:45 -0500 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. Mike p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: > Doing the building thing, seem to have come across a bug. > > Have a look at Part 2 of http://makingfs.blogspot.com/ > > If make crashes out, it states that it was successfully built ;) > > Julian > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From asterisk at dotr.com Fri Dec 11 10:47:13 2009 From: asterisk at dotr.com (Julian Lyndon-Smith) Date: Fri, 11 Dec 2009 18:47:13 +0000 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: Thanks Mike. I understand why you don't want all to be built. However, there are things that I would like - such as mod_java. However, that fails to compile, I presume because of some missing dependency or requirement. Is there any tool to tell me what is needed in order to build a module ? Julian 2009/12/11 Michael Jerris : > It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. ?The basics of this is that automake continues on to other subdirs if build in one subdir fails. > > Mike > > p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. ?This module is barely touched and I would not use it unless you have a good reason to. > > > On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: > >> Doing the building thing, seem to have come across a bug. >> >> Have a look at Part 2 of http://makingfs.blogspot.com/ >> >> If make crashes out, it states that it was successfully built ;) >> >> Julian >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Fri Dec 11 11:03:58 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:03:58 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> Message-ID: <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Hi Mike, the fs console log with sip trace on the internal profile is attached in the pastebin below, http://pastebin.freeswitch.org/11483 could you please take a look at it? Thanks, Chris On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris wrote: > As i said multiple times on irc last night, we need to see debug logs with > sip trace to see what is going on. > > Mike > > On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > > Thanks Frank for sharing your experience. This is the behavior change just > starting within three days, maybe because of some code changes in mod_sofia > which I should change the settings accordingly > I noticed that the acl automatically having 192.168.0.0 set as "deny", > that's why I tried to changed the settings regarding nat acl and localnet > acl. > > Chris > > > > On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: > >> On Fri, Dec 11, Chris Chen wrote: >> > Hi there, I have very strange behaviors for my SIP endpoints with FS SVN >> > trunk 15905. >> >> Is this a change in behavior or is this the first time you've run >> freeswitch? If this is your first time welcome aboard! Also if this is >> your first time you've probably have some IPs aliased on your interface and >> you still have stun enabled. This was the behavior I saw the first time I >> ran it on a box with aliases on an interface. The stun server tells >> freeswitch after some time that the IP is different then the one you've >> assigned. This is just one possibility. If this isn't the case then we >> will need to see sip traces on all of your profiles. >> >> --FC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/432df45f/attachment-0002.html From mrene_lists at avgs.ca Fri Dec 11 11:09:06 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 11 Dec 2009 14:09:06 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Message-ID: Its not sending to the right Contact: header in the 200 OK packet. This was fixed in r15870, you have to update. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Dec-09, at 2:03 PM, Chris Chen wrote: > Hi Mike, the fs console log with sip trace on the internal profile > is attached in the pastebin below, > http://pastebin.freeswitch.org/11483 > > could you please take a look at it? > Thanks, > Chris > > On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris > wrote: > As i said multiple times on irc last night, we need to see debug > logs with sip trace to see what is going on. > > Mike > > On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: > >> Thanks Frank for sharing your experience. This is the behavior >> change just starting within three days, maybe because of some code >> changes in mod_sofia which I should change the settings accordingly >> I noticed that the acl automatically having 192.168.0.0 set as >> "deny", that's why I tried to changed the settings regarding nat >> acl and localnet acl. >> >> Chris >> >> >> >> On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle > > wrote: >> On Fri, Dec 11, Chris Chen wrote: >> > Hi there, I have very strange behaviors for my SIP endpoints with >> FS SVN >> > trunk 15905. >> >> Is this a change in behavior or is this the first time you've run >> freeswitch? If this is your first time welcome aboard! Also if >> this is your first time you've probably have some IPs aliased on >> your interface and you still have stun enabled. This was the >> behavior I saw the first time I ran it on a box with aliases on an >> interface. The stun server tells freeswitch after some time that >> the IP is different then the one you've assigned. This is just one >> possibility. If this isn't the case then we will need to see sip >> traces on all of your profiles. >> >> --FC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/d467295a/attachment-0002.html From mike at jerris.com Fri Dec 11 11:11:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Dec 2009 14:11:37 -0500 Subject: [Freeswitch-users] The Building Freeswitch blog In-Reply-To: References: Message-ID: <09EB5512-D5EC-489C-9C47-9BB3A8F6A502@jerris.com> Probably the best list is: http://wiki.freeswitch.org/wiki/FreeSwitch_Dependencies Due to the fact that we allow you to change modules after configure there is no great way to have it error out when you don't have the right deps other than to just have the compile errors when you try to build. Its probably time for a tool like make menuconfig but we do not have that as of yet. Mike On Dec 11, 2009, at 1:47 PM, Julian Lyndon-Smith wrote: > Thanks Mike. I understand why you don't want all to be built. However, > there are things that I would like - such as mod_java. However, that > fails to compile, I presume because of some missing dependency or > requirement. Is there any tool to tell me what is needed in order to > build a module ? > > Julian > > 2009/12/11 Michael Jerris : >> It just so happens I was looking at this same bug last night and having troubles chasing down a solution, if anyone comes up with anything good please let me know. The basics of this is that automake continues on to other subdirs if build in one subdir fails. >> >> Mike >> >> p..s. a note on the blog, I generally do not recommend just building everything, for example, mod_alsa is a module written specifically for the n800 due to mod_portaudio not working there. This module is barely touched and I would not use it unless you have a good reason to. >> >> >> On Dec 11, 2009, at 1:19 PM, Julian Lyndon-Smith wrote: >> >>> Doing the building thing, seem to have come across a bug. >>> >>> Have a look at Part 2 of http://makingfs.blogspot.com/ >>> >>> If make crashes out, it states that it was successfully built ;) >>> >>> Julian >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris.chen2004 at gmail.com Fri Dec 11 11:34:26 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:34:26 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> Message-ID: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Thanks Mathieu, but I am on SVN r15912 now. Chris On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene wrote: > Its not sending to the right Contact: header in the 200 OK packet. This > was fixed in r15870, you have to update. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 11-Dec-09, at 2:03 PM, Chris Chen wrote: > > Hi Mike, the fs console log with sip trace on the internal profile is > attached in the pastebin below, > http://pastebin.freeswitch.org/11483 > > could you please take a look at it? > Thanks, > Chris > > On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris wrote: > >> As i said multiple times on irc last night, we need to see debug logs with >> sip trace to see what is going on. >> >> Mike >> >> On Dec 11, 2009, at 11:44 AM, Chris Chen wrote: >> >> Thanks Frank for sharing your experience. This is the behavior change just >> starting within three days, maybe because of some code changes in mod_sofia >> which I should change the settings accordingly >> I noticed that the acl automatically having 192.168.0.0 set as "deny", >> that's why I tried to changed the settings regarding nat acl and localnet >> acl. >> >> Chris >> >> >> >> On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle wrote: >> >>> On Fri, Dec 11, Chris Chen wrote: >>> > Hi there, I have very strange behaviors for my SIP endpoints with FS >>> SVN >>> > trunk 15905. >>> >>> Is this a change in behavior or is this the first time you've run >>> freeswitch? If this is your first time welcome aboard! Also if this is >>> your first time you've probably have some IPs aliased on your interface and >>> you still have stun enabled. This was the behavior I saw the first time I >>> ran it on a box with aliases on an interface. The stun server tells >>> freeswitch after some time that the IP is different then the one you've >>> assigned. This is just one possibility. If this isn't the case then we >>> will need to see sip traces on all of your profiles. >>> >>> --FC >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/756d2adc/attachment-0002.html From kristian.kielhofner at gmail.com Fri Dec 11 11:41:19 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 11 Dec 2009 14:41:19 -0500 Subject: [Freeswitch-users] bridge doesn't respect bypass_media=true over the socket In-Reply-To: <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> References: <2d9149cd0912110914w3814651dqa943f000fc174fb3@mail.gmail.com> <191c3a030912110931n595961e1wa3d1d496ee1c8c3b@mail.gmail.com> Message-ID: <2d9149cd0912111141r575be558s14d2002a9cee1d6e@mail.gmail.com> Thanks, that was it! On Fri, Dec 11, 2009 at 12:31 PM, Anthony Minessale wrote: > Hey, > > You can't set bypass_media=true in {} or it will not take effect unless that > b leg itself becomes an a leg some day. > you need to execute set on bypass_media=true on the leg before you call > bridge to trigger it. > > Alternatively you could set {bypass_media_after_bridge=true} or set it on A > leg as described above on either leg and it will do the bypass once the > audio is flowing. > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Fri Dec 11 11:45:23 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 13:45:23 -0600 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Message-ID: <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> You set the extrtp ip to an IP exactly.. this is the issue we are fixing soon.. if you have natpmp or upnp set it to auto-nat and let it figure it out. The issue is we have restored the behavior in 1.0.4 that lies about the IP all the time... I'm going to commit a patch shortly that'll fix this. /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris From brian at freeswitch.org Fri Dec 11 11:56:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 13:56:49 -0600 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> Message-ID: Please test www.bkw.org/sofia_autonat_static_ip.diff /b On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > Thanks Mathieu, but I am on SVN r15912 now. > > Chris From chris.chen2004 at gmail.com Fri Dec 11 11:57:49 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Fri, 11 Dec 2009 14:57:49 -0500 Subject: [Freeswitch-users] Audio cut after 31 seconds for SIP endpoints with latest FS SVN trunks In-Reply-To: <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> References: <507898380912110806i13835153vb6802c054ade0c86@mail.gmail.com> <20091211162459.GR31924@base.carmickle.com> <507898380912110844l54d72bd4mca10e384345d15d6@mail.gmail.com> <800D2722-1FF7-4262-B89D-0826B5E64049@jerris.com> <507898380912111103t1b743cffv66c9859eee9aee50@mail.gmail.com> <507898380912111134g1406a65ale222e2be20751f30@mail.gmail.com> <6E780C71-DC56-4B19-BF87-8B3D4F4AD996@freeswitch.org> Message-ID: <507898380912111157r56f88862ua498166904aed2b4@mail.gmail.com> Thanks Brian for your explanation, could we still keep the option to set the extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe many other routers have similar issue. Chris On Fri, Dec 11, 2009 at 2:45 PM, Brian West wrote: > You set the extrtp ip to an IP exactly.. this is the issue we are > fixing soon.. if you have natpmp or upnp set it to auto-nat and let it > figure it out. The issue is we have restored the behavior in 1.0.4 > that lies about the IP all the time... > > I'm going to commit a patch shortly that'll fix this. > > /b > > On Dec 11, 2009, at 1:34 PM, Chris Chen wrote: > > > Thanks Mathieu, but I am on SVN r15912 now. > > > > Chris > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/7f1c2a9d/attachment-0002.html From bcxml at hotmail.com Fri Dec 11 15:11:31 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 15:11:31 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound Message-ID: <26752894.post@talk.nabble.com> I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass an incomming PSTN call to Speech Server. But I cannot get Freeswitch to dial out a call or transfer a call that is sent from Speech Server I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called voipms.xml which contains the following..(I have an account with voip.ms) And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called Outbound.xml which contains the following When my Speech Server application tries to get FreeSwitch to transfer to another number, the console shows the following 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/+1 9059183027 at 199.173.95.16:5060 to XML[%23904161234 at public] 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/interna l/2482578002 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing +14165551212 ->%23904161234 in context public 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/ internal/2482578002 at 127.0.0.1:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sof ia/internal/2482578002 at 127.0.0.1:5060 [CS_DESTROY] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup sofia /external/+19059183027 at 199.173.95.16:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/ external/+19059183027 at 199.173.95.16:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sof ia/external/+19059183027 at 199.173.95.16:5060 [CS_DESTROY] I really dont understand the line above that I have in Bold & Italic The number being transfered to was 4161234567...so I would have thought the line should read.. Processing +14165551212->4161234567 in context public Can anyone tell me what the "%2390" means and also any problems with my XML files that could be preventing the transfers from taking place Thanks Brian -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26752894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From bcxml at hotmail.com Fri Dec 11 15:23:19 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 15:23:19 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound Message-ID: <26752894.post@talk.nabble.com> I am very new to Freeswitch so please accept my appologies if these questions seem to be trivial I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. I have been successful in getting Freeswitch to pass an incomming PSTN call to Speech Server. But I cannot get Freeswitch to dial out a call or transfer a call that is sent from Speech Server I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called voipms.xml which contains the following..(I have an account with voip.ms) And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called Outbound.xml which contains the following When my Speech Server application tries to get FreeSwitch to transfer to another number, the console shows the following 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/+14052582456 at 111.222.333.444:5060 to XML[%23904161234 at public] 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup sofia/internal/2484487788 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing +14052582456->%23904161234 in context public 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/internal/2484487788 at 127.0.0.1:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/2484487788 at 127.0.0.1:5060 [CS_DESTROY] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/+14052582456 at 111.222.333.444:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/+14052582456 at 111.222.333.444:5060) Ended 2009-12-11 17:54:31.879138 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/+14052582456 at 111.222.333.444:5060 [CS_DESTROY] I really dont understand the line above that I have in Bold & Italic The number being transfered to was 4161234567...so I would have thought the line should read.. "Processing +14052582456->4161234567 in context public " Can anyone tell me what the "%2390" means and also any problems with my XML files that could be preventing the transfers from taking place Thanks Brian -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26752894.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Fri Dec 11 15:34:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Dec 2009 15:34:14 -0800 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <26752894.post@talk.nabble.com> References: <26752894.post@talk.nabble.com> Message-ID: <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> On Fri, Dec 11, 2009 at 3:11 PM, bcxml wrote: > > I am very new to Freeswitch so please accept my appologies if these > questions > seem to be trivial > > I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007. > I > have been successful in getting Freeswitch to pass an incomming PSTN call > to > Speech Server. But I cannot get Freeswitch to dial out a call or transfer a > call that is sent from Speech Server > > I have a file setup in C:\FreeSWITCH\conf\sip_profiles\external\ called > voipms.xml which contains the following..(I have an account with voip.ms) > > > > > > > > > > > > And I have a file setup in C:\FreeSWITCH\conf\dialplan\public\ called > Outbound.xml which contains the following > > > > data="effective_caller_id_number=12223334444"/> > > > > > When my Speech Server application tries to get FreeSwitch to transfer to > another number, the console shows the following > > > 2009-12-11 17:54:31.863512 [NOTICE] switch_ivr.c:1349 Transfer > sofia/external/+1 > 9059183027 at 199.173.95.16:5060 to XML[%23904161234 at public] > 2009-12-11 17:54:31.879138 [NOTICE] switch_ivr_bridge.c:503 Hangup > sofia/interna > l/2482578002 at 127.0.0.1:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > > 2009-12-11 17:54:31.879138 [INFO] mod_dialplan_xml.c:315 Processing > +14165551212 > ->%23904161234 in context public > This line is basically saying that you have a call coming from 4165551212 and it's looking for a destination number of %23904161234. The key here is that it is coming in the public context so you'll need to handle the routing in conf/dialplan/public.xml What should this call be doing once it comes in to FS? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091211/c85d280d/attachment-0002.html From codecomplete at free.fr Fri Dec 11 15:38:19 2009 From: codecomplete at free.fr (Fred-145) Date: Fri, 11 Dec 2009 15:38:19 -0800 (PST) Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> Message-ID: <26753168.post@talk.nabble.com> Right, when talking about NAT firewall, I meant the outer NAT, not the one that could be running on the same host where FS is installed. I'll see if I can find a utility that checks that the ports are open after FS is up and running. Thank you. -- View this message in context: http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26753168.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Dec 11 15:43:33 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 17:43:33 -0600 Subject: [Freeswitch-users] Does FS support STUN by default? In-Reply-To: <26753168.post@talk.nabble.com> References: <26727762.post@talk.nabble.com> <26731188.post@talk.nabble.com> <0E791F13-DF48-4916-96C3-1CE50F587375@jerris.com> <26740589.post@talk.nabble.com> <26748901.post@talk.nabble.com> <30340A58-DD2C-403F-9834-1F99DDD94072@freeswitch.org> <26753168.post@talk.nabble.com> Message-ID: <4C923D5E-8A20-4905-B917-F45F72300343@freeswitch.org> You don't have to do that usually... /b On Dec 11, 2009, at 5:38 PM, Fred-145 wrote: > I'll see if I can find a utility that checks that the ports are open > after > FS is up and running. From brian at freeswitch.org Fri Dec 11 15:44:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Dec 2009 17:44:49 -0600 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> Message-ID: %23 is # so the question is should we URL decode that before routing? I thought we did... what version are you using now? /b On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: > This line is basically saying that you have a call coming from > 4165551212 and it's looking for a destination number of > %23904161234. The key here is that it is coming in the public > context so you'll need to handle the routing in conf/dialplan/ > public.xml > > What should this call be doing once it comes in to FS? > > -MC From bcxml at hotmail.com Fri Dec 11 16:02:16 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 11 Dec 2009 16:02:16 -0800 (PST) Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> Message-ID: <26753346.post@talk.nabble.com> The version is FreeSWITCH Version 1.0.4 (14460) Brian Brian West-3 wrote: > > %23 is # so the question is should we URL decode that before routing? > I thought we did... what version are you using now? > > /b > > On Dec 11, 2009, at 5:34 PM, Michael Collins wrote: > >> This line is basically saying that you have a call coming from >> 4165551212 and it's looking for a destination number of >> %23904161234. The key here is that it is coming in the public >> context so you'll need to handle the routing in conf/dialplan/ >> public.xml >> >> What should this call be doing once it comes in to FS? >> >> -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Problems-with-Freeswitch-setup---Outbound-tp26752894p26753346.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Fri Dec 11 22:14:23 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 11 Dec 2009 22:14:23 -0800 Subject: [Freeswitch-users] Problems with Freeswitch setup - Outbound In-Reply-To: <26753346.post@talk.nabble.com> References: <26752894.post@talk.nabble.com> <87f2f3b90912111534y266a57f0j3568792facf3b31b@mail.gmail.com> <26753346.post@talk.nabble.com> Message-ID: On Dec 11, 2009, at 4:02 PM, bcxml wrote: > > The version is > > FreeSWITCH Version 1.0.4 (14460) > Ouch. You are nearly 6 months and 1500 revs behind. You badly need to update to latest trunk. -MC > >> >> >> >>> >>> >> >> From thangappan143 at gmail.com Fri Dec 11 23:38:51 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 12 Dec 2009 13:08:51 +0530 Subject: [Freeswitch-users] Fwd: Problem while playing more than 10 voice files using playback In-Reply-To: <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> References: <7aa29e790911210122t604fbfd5mf2ae8235fe83e6d3@mail.gmail.com> <7aa29e790911222034x3d8159abm1e156beb1738c8ac@mail.gmail.com> <7aa29e790911232156w6c2acc93l78666dd6575e0efb@mail.gmail.com> <7aa29e790911242209w7ee2912bhbde4b3475147628d@mail.gmail.com> <7aa29e790911242218l580b90eem3ec50676dfbc5536@mail.gmail.com> Message-ID: <7aa29e790912112338g23ab5867qeeca426192f34709@mail.gmail.com> Thanks for your answers. Rather than select or polling the variable on the channel, CHANNEL EXECUTE COMPLETE event is used and now it is being worked fine. ---------- Forwarded message ---------- From: Thangappan.M Date: Wed, Nov 25, 2009 at 11:48 AM Subject: Re: Problem while playing more than 10 voice files using playback To: freeswitch-users The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M FreeSWITCH version: freeswitch 1.0.4 > I am using ESL library > I attached the example Perl script which does the same steps that I posted > already. ( Sample.pl) > I supplied the log , Here I attached the output of the ESL log. > (Output.txt) > > Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. > But in the output I got only 2,4,5,4 ( DTMF 1 is missed) > > Output of Perl code could be like > > Wait for response time out > EVENT [COMMAND] > Wait for response time out > EVENT [DTMF] > DTMF digit 2 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 5 (2000) > Wait for inter digit time out > EVENT [DTMF] > DTMF digit 4 (2000) > Wait for inter digit time out > Buffer: 2454 > BYE > > Why the first digit(1) is missed here? > In ESL log there is no digit called 1 why? > Why the COMMAND event is received instead of DTMF? > How can I get all DTMF digits? > > > > > > > > > > > > > > > > > On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M wrote: > >> The reason for waiting only for DTMF event is to handle the time outs in >> the IVR concept like response and inter digit time out. Using our own logic >> we 10 voice files in each play back if the voice files are more than 10. Now >> it works fine. >> >> Now the new problem has been raised. The problem is we are filtering only >> for DTMF events but we are getting COMMAND event . Because of this the DTMF >> digits are missing at the time . I am not able to proceed further. We are >> in the critical situation. >> >> Why this command event is occurring? >> How can I restrict this? >> What are the information it has? >> How can I get all the information in it ? ( If command event has info) >> >> Help me............ >> >> >> On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M wrote: >> >>> I am waiting only for DTMF events. That's why I am setting freeswitch >>> variable for knowing whether the playback has done. >>> >>> My question is "why this freeswitch variable is not setting properly when >>> I play back more than 10 files using playback_delimiter option?". >>> >>> When I play back lesser than ten voice files the variable has been set >>> properly. What could be the reason? >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Thangappan.M >>> Date: Sat, Nov 21, 2009 at 2:52 PM >>> Subject: Problem while playing more than 10 voice files using playback >>> To: freeswitch-users >>> >>> >>> Dear all, >>> >>> I am in the process of implementing IVR using event outbound >>> socket (async mode). >>> I have implemented using Perl language. >>> >>> I did the following steps: >>> => Set the playback_delimiter variable >>> => Set the playback_sleep_val variable >>> => Set the event lock as true >>> => Set the freeswitch ( my own) variable as zero >>> => Wait in the loop until the variable is been set as >>> zero >>> => Playback the voice files ( Here I combined the >>> voice files with the delimiter value if more than one voice files are there) >>> => Set the freeswitch(my own) variable as true ( This >>> is used to identify whether the voice files are played >>> successfully). >>> => Wait in the loop until the variable is been set as >>> one. >>> => Set the Event lock as false >>> >>> => Trying to get the DTMF digits ( Have a assurance >>> that all the voice files are played). >>> >>> The problem is, >>> >>> The above steps are working fine when the voice file count >>> is lesser than or equal to 10. After the voice files are played only the >>> variable(my own freeswitch) is set. Based on the variable I am doing further >>> things. >>> >>> But when I tried to give the voice files count of more than >>> 10 the variable has been set while starting to play back the first voice >>> file itself . Because of this I am not able to proceed further. >>> >>> *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* >>> >>> *NOTE*: I also referred mod_file_string documentation. In that they >>> specified 128 files can be used to play back the voice files using >>> playback_delimiter option. >>> >>> Please help me................? >>> Thanks in advance. >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >>> >>> >>> -- >>> Regards, >>> Thangappan.M >>> >> >> >> >> -- >> Regards, >> Thangappan.M >> > > > > -- > Regards, > Thangappan.M > -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/b2a96371/attachment-0002.html From thangappan143 at gmail.com Fri Dec 11 23:42:23 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 12 Dec 2009 13:12:23 +0530 Subject: [Freeswitch-users] Getting started on IVR Library Message-ID: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me....... -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/ae3f7a7c/attachment-0002.html From surajee at gmail.com Fri Dec 11 23:08:13 2009 From: surajee at gmail.com (Surajee Ratnayake) Date: Sat, 12 Dec 2009 12:38:13 +0530 Subject: [Freeswitch-users] Freeswitch and Gtalk Message-ID: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Hello.. just want to get the following clarified from the friends in the same domain, since freeswitch is allowing multiple gtalk user registrations with gtalk servers, assume we route gtalk voice calls coming to these gtalk users are routed to sip extensions or to PSTN/PLMN? will google block some thing like that or is it already happening? scenario is, gtalk client A----------------------> gtalk user B at Freeswitch ------------------------------> PSTN thanx in advance, Sur -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/af3e514f/attachment-0002.html From mike at jerris.com Sat Dec 12 00:50:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 12 Dec 2009 03:50:47 -0500 Subject: [Freeswitch-users] Getting started on IVR Library In-Reply-To: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Message-ID: A good example of how to use this code would be in mod_rss or mod_voicemail in tree. I would say look at the doxygen at http://docs.freeswitch.org/group__switch__ivr.html but it appears that page is completely broken. I will try to take a look and figure out why this weekend, in the meantime, you can look at the doxygen comments inline in switch_ivr*.h. Mike On Dec 12, 2009, at 2:42 AM, Thangappan.M wrote: > Dear all , > > I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/8f2564c9/attachment-0002.html From mike at jerris.com Sat Dec 12 01:01:51 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 12 Dec 2009 04:01:51 -0500 Subject: [Freeswitch-users] Freeswitch and Gtalk In-Reply-To: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Message-ID: That should work fine. Mike On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote: > Hello.. > just want to get the following clarified from the friends in the same domain, > > since freeswitch is allowing multiple gtalk user registrations with gtalk servers, assume we route gtalk voice calls coming to these gtalk users are routed to sip extensions or to PSTN/PLMN? will google block some thing like that or is it already happening? From mailinglist at fribert.dk Sat Dec 12 03:28:37 2009 From: mailinglist at fribert.dk (mailinglist) Date: Sat, 12 Dec 2009 12:28:37 +0100 Subject: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through In-Reply-To: <117186.49301.qm@web56406.mail.re3.yahoo.com> References: <4B213D87020000E100000322@mail.fribert.dk> <117186.49301.qm@web56406.mail.re3.yahoo.com> Message-ID: <4B238C75020000E10000032D@mail.fribert.dk> Hi Mark et al. Don't know where my reply went, but certainly not to the list :-) So my public: default.xml looks like this; If I don't have any phones externally to my FS, could I just empty the public's default.xml? BR Fribert >>> 11-12-2009 kl. 10:44 skrev Mark Crane i meddelelsen <117186.49301.qm at web56406.mail.re3.yahoo.com>: What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Thu, 12/10/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Thursday, December 10, 2009, 10:27 AM Hi Mark The extensions are located under directory/default, and they look like this: As I understand them, the context set there, is the right one? >>> 10-12-2009 kl. 00:00 skrev Mark Crane i meddelelsen <110796.60596.qm at web56401.mail.re3.yahoo.com>: Please check both extensions and make sure that the 'User Context' is set to: default The dialplan you showed has this. Which finds the destination_number of the extension you are calling and then sends it there. But from the logs you showed earlier it did not make it this far in the dialplan. You need to find out where its getting diverted. The strange thing is I can see it goes into the dialplan and starts making the comparison to the regular expressions compares two or three then moves on without a match which isn't standard behavior. Some of what I read hints toward is running on the public interface (external) when calling. What do you have for the 'public' tab and is there any entries with anti-actions? Mark --- On Wed, 12/9/09, mailinglist wrote: From: mailinglist Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users at lists.freeswitch.org Date: Wednesday, December 9, 2009, 3:20 PM WARNING LONG POST! It must be something I've done wrong with the dialplan setup but I can't find any of the statements in the default, that should grab the call? In conf/dialplan I have default (dir) default.xml features.xml public (dir) public.xml The default.xml looks like this ( I haven't changed it): ]]> Then I have under default dir: musimidk.xml and 9000_recordings.xml Somewhere in that very long default dialplan the call must be directed out. As I can understand the default.xml it includes every file that matches *.xml under the dialplan directory, so it should pick it up??? -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org ( /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org ) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/482f7ecf/attachment-0002.html From dftoro at yahoo.com Sat Dec 12 10:01:22 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sat, 12 Dec 2009 10:01:22 -0800 (PST) Subject: [Freeswitch-users] fifo caller id In-Reply-To: <4B238C75020000E10000032D@mail.fribert.dk> Message-ID: <319374.69139.qm@web33508.mail.mud.yahoo.com> Hello, ? I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and? fifo_consumer_caller_import variables, but i don't know use it. ? I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/b9e0539e/attachment-0002.html From surajee at gmail.com Sat Dec 12 14:34:48 2009 From: surajee at gmail.com (Surajee Ratnayake) Date: Sun, 13 Dec 2009 04:04:48 +0530 Subject: [Freeswitch-users] Freeswitch and Gtalk In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> Message-ID: <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> Thank you, I believe that will legally be okay for google also. I was worried that they might block access to our servers later Sur On 12/12/09, Michael Jerris wrote: > That should work fine. > > Mike > > On Dec 12, 2009, at 2:08 AM, Surajee Ratnayake wrote: > >> Hello.. >> just want to get the following clarified from the friends in the same >> domain, >> >> since freeswitch is allowing multiple gtalk user registrations with gtalk >> servers, assume we route gtalk voice calls coming to these gtalk users are >> routed to sip extensions or to PSTN/PLMN? will google block some thing >> like that or is it already happening? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From abeka at greatiam.com Sat Dec 12 15:17:46 2009 From: abeka at greatiam.com (Otis) Date: Sat, 12 Dec 2009 23:17:46 +0000 Subject: [Freeswitch-users] Link between User context and dialplan Message-ID: <4B24249A.1040004@greatiam.com> Hi folks I am so sorry if this is such a basic thing. well, when a user/extension eg 8888 is created in with say a user context - SWAHILI-SPEAKERS Please hear are my questions: 1. What dialplan will that user/extn use. 2. I guess I have to create a dialplan Should the dial-plan also be called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be any name ? 3. And how does FS know to load that dialplan for that user. 4. Where should that xml file be stored ? 5. Is there a means of determinig which dialplan was used for a call ? Thanks I think I have demonstrated enough thickness for now From dfansler at dv-fansler.com Sat Dec 12 19:28:14 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sat, 12 Dec 2009 22:28:14 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> Message-ID: <01a401ca7ba4$4e449190$eacdb4b0$@com> I am new to FreeSWITCH (ok a month old) and am still learning as hard as I can. In the recent talk about documentation, I had noticed that finding documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to download all the wiki pages and put them in some sort of logical order. No I am not finished yet, but does anyone else out there realize how many wiki pages there are on FreeSWITCH's site? So far I am up to 767 pages and have another 100 to go. I had no idea what I was getting into! David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From abeka at greatiam.com Sat Dec 12 19:40:08 2009 From: abeka at greatiam.com (Otis) Date: Sun, 13 Dec 2009 03:40:08 +0000 Subject: [Freeswitch-users] Link between Use-context and dialplan Message-ID: <4B246218.2020804@greatiam.com> Sorry I posted this earlier but did not do the due diligence and sent it with so much typo them meaning does not come out: In a nutshell I would like to know : 1. How FS would know which dialplan to use for an extension with user context other than default. 2. If a file file has to be created does the name matter 3. Where should that file be located. Thanks. From rjcajax at gmail.com Sat Dec 12 19:41:54 2009 From: rjcajax at gmail.com (Robert Clayton) Date: Sat, 12 Dec 2009 22:41:54 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <01a401ca7ba4$4e449190$eacdb4b0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: David, Thanks for your hard work. Maybe more organization will make the areas needing substance or explanation more obvious. Bob On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/01b891cb/attachment-0002.html From msc at freeswitch.org Sat Dec 12 19:47:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Dec 2009 19:47:19 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <01a401ca7ba4$4e449190$eacdb4b0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: <343FDCB3-6D21-4D0D-B51E-8BC2C57B202F@freeswitch.org> On Dec 12, 2009, at 7:28 PM, "David V. Fansler" wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as > hard as I > can. In the recent talk about documentation, I had noticed that > finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I > decided to > download all the wiki pages and put them in some sort of logical > order. No > I am not finished yet, but does anyone else out there realize how > many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages > and have > another 100 to go. > > I had no idea what I was getting into! > Hehe neither did I when I started a few years ago. This is a monumetal task but since you've overcome the inertia let's just roll up our sleeves and get it done. I'm glad to have your help. I will contact you off list to follow up. Thanks! -MC (IRC: mercutioviz) > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Sat Dec 12 19:48:51 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Dec 2009 19:48:51 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: On Dec 12, 2009, at 7:41 PM, Robert Clayton wrote: > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. That's my hope as well. -MC > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler > wrote: > I am new to FreeSWITCH (ok a month old) and am still learning as > hard as I > can. In the recent talk about documentation, I had noticed that > finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I > decided to > download all the wiki pages and put them in some sort of logical > order. No > I am not finished yet, but does anyone else out there realize how > many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages > and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091212/e8dd5498/attachment-0002.html From help at pdscc.com Sun Dec 13 01:53:43 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Sun, 13 Dec 2009 01:53:43 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20090823235030.A95BF5FE@sinclaire.sibble.net>, <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org> Message-ID: <20091213095350.F01371646@sinclaire.sibble.net> Brian, you have any inside info on when zfone3 is supposed to be available? Emails to the zfoneproject addresses haven't gotten me any info (or a reponse for that matter), I guess cause I am not a dev... On 23 Aug 2009 at 18:53, Brian West wrote: > Wish they would send me one for my E63 for testing... only been > working with zfone 3 so far. > > /b > > On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote: > > > Well good news for the Tiviphone client -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From dujinfang at gmail.com Sun Dec 13 04:54:53 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 13 Dec 2009 20:54:53 +0800 Subject: [Freeswitch-users] fifo caller id In-Reply-To: <319374.69139.qm@web33508.mail.mud.yahoo.com> References: <4B238C75020000E10000032D@mail.fribert.dk> <319374.69139.qm@web33508.mail.mud.yahoo.com> Message-ID: <23f91030912130454w21319313i82b21b2633cad2e5@mail.gmail.com> I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro > Hello, > > I want to know how can I get caller id after call is out queue fifo, I read > about fifo_caller_consumer_import and fifo_consumer_caller_import > variables, but i don't know use it. > > I appreciate any suggestion > > Diego Toro > http://lacarretade.blogspot.com/ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/41a3913e/attachment-0002.html From jmesquita at freeswitch.org Sun Dec 13 06:46:36 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 13 Dec 2009 12:46:36 -0200 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <4B246218.2020804@greatiam.com> References: <4B246218.2020804@greatiam.com> Message-ID: inline.... JM On Sun, Dec 13, 2009 at 1:40 AM, Otis wrote: > Sorry > > I posted this earlier but did not do the due diligence and sent it with > so much typo them meaning does not come out: > > In a nutshell I would like to know : > > 1. How FS would know which dialplan to use for an extension with user > context other than default. > The SIP profile that the call comes in has a context. All calls that do not have users associated (not authenticated) or users that do not have the user_context var set will use that context. If the user has the user_context var set, it will use the specified one. > 2. If a file file has to be created does the name matter > No. 3. Where should that file be located. > > ${FSROOT}/conf/dialplan/* I *strongly *suggest you to read the default configs and the wiki. > Thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/b54a00b6/attachment-0002.html From yehavi.bourvine at gmail.com Sun Dec 13 06:51:04 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 16:51:04 +0200 Subject: [Freeswitch-users] Sofia performance Message-ID: Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/71604c28/attachment-0002.html From tzury.by at reguluslabs.com Sun Dec 13 06:56:26 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Sun, 13 Dec 2009 09:56:26 -0500 Subject: [Freeswitch-users] sangoma a101 - error compiling Wanpipe drivers with TDM API support on ubuntu 9.10 Message-ID: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> Hi all, I am getting a frustrating errors while trying to build the latest wanpipe on my newly and freshly installed Ubuntu 9.10 I pastebin the output at http://gist.github.com/255442 and would appreciate any help. Thanks, -- Tzury Bar Yochay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/5acb7003/attachment-0002.html From niall.crosby at gmail.com Sun Dec 13 07:05:12 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 15:05:12 +0000 Subject: [Freeswitch-users] Java ESL Message-ID: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Hi, I am about to start writing a Java Event Socket Library as I can't find one already written thats available. 1 - Is there one already out there? 2 - If not, any pointers as to what design I should follow? Which of the current ESL's is the best modal to follow? Thanks, Niall. -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/396134a7/attachment-0002.html From jbr at consiglia.dk Sun Dec 13 07:19:46 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Sun, 13 Dec 2009 16:19:46 +0100 Subject: [Freeswitch-users] Presence across several networked FSs Message-ID: I'm working on a setup with several networked FreeSWITCHes based on a central FS and one or more satellite FSs. The boxes are connected to each other with IAX, SIP over VPN or another protocol. In the choice of the protocol the presence issue mentioned below should be considered, and the real life practicalities such as routers, NAT and connections with packet loss. I find the directory facility as a good tool for expressing the topology on each server, where the dial-string in the directory can be used to beak out of the box into some of the other boxes via the central unit. When I dial out from a phone, the presence information is updated OK on same phone and other phones on the same box. I would like to keep track of the presence of the users on the other boxes as well. Any suggestion on to how the presence information scan be propagated to all boxes in the network. Jon Br?el Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/41ea0c39/attachment-0002.html From frank at carmickle.com Sun Dec 13 07:21:59 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 10:21:59 -0500 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <4B246218.2020804@greatiam.com> References: <4B246218.2020804@greatiam.com> Message-ID: <20091213152158.GV31924@base.carmickle.com> On Sun, Dec 13, Otis wrote: > Sorry > > I posted this earlier but did not do the due diligence and sent it with > so much typo them meaning does not come out: > > In a nutshell I would like to know : > > 1. How FS would know which dialplan to use for an extension with user > context other than default. It just uses the context tag which you include extensions in side it. > 2. If a file file has to be created does the name matter Doesn't matter. The include statements pull the files in. Wrap the stuff in the included file in tags. > 3. Where should that file be located. Anywhere! It easiest to set up includes like Now you can put all kinds of files in the public dir and they will get included when the preprocess runs. The preprocess runs at start up so you need to restart IIRC. HTH --FC From jmesquita at freeswitch.org Sun Dec 13 07:39:06 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 13 Dec 2009 13:39:06 -0200 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Message-ID: Can't we just swig it to Java? JM On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby wrote: > > Hi, > > I am about to start writing a Java Event Socket Library as I can't find one > already written thats available. > 1 - Is there one already out there? > 2 - If not, any pointers as to what design I should follow? Which of the > current ESL's is the best modal to follow? > > Thanks, > Niall. > > -- > -- > > The information transmitted is intended only for the person or entity to > which it is addressed and may contain confidential and/or privileged > material. Statements and opinions expressed in this e-mail may not represent > those of the sender. Any review, retransmission, dissemination or other use > of, or taking of any action in reliance upon, this information by persons or > entities other than the intended recipient is prohibited. If you received > this in error, please contact the sender immediately and delete the material > from any computer. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/2628e43a/attachment-0002.html From Russell.Mosemann at cune.org Sun Dec 13 07:44:16 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Sun, 13 Dec 2009 09:44:16 -0600 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <20091213152158.GV31924@base.carmickle.com> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> Message-ID: <115561884BFA43AAA5CACA534551E5F1@cune.pri> Frank Carmickle wrote: > Now you can put all kinds of files in the public dir and they will get > included when the preprocess runs. The preprocess runs at start up so you > need to restart IIRC. Or fire up /bin/fs_cli and issue a "reloadxml" or "reload ", if a change affects a module. -- Russell Mosemann From dujinfang at gmail.com Sun Dec 13 07:47:04 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 13 Dec 2009 23:47:04 +0800 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: Message-ID: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> you can use the same ip with different port 2009/12/13, Yehavi Bourvine : > Hello, > > In the WIKI page that talks about Freeswitch performance there is a > sentence: > > *libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles* > > How can I enable more than one profile on the same interface? Won't they > colide when using the same IP and port? > > Thanks! __Yehavi: > From niall.crosby at gmail.com Sun Dec 13 07:50:28 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 15:50:28 +0000 Subject: [Freeswitch-users] Java ESL In-Reply-To: References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> Message-ID: <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> Pure Java is my preference - am looking to build apps that are portable. N. 2009/12/13 Jo?o Mesquita > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby wrote: > >> >> Hi, >> >> I am about to start writing a Java Event Socket Library as I can't find >> one already written thats available. >> 1 - Is there one already out there? >> 2 - If not, any pointers as to what design I should follow? Which of the >> current ESL's is the best modal to follow? >> >> Thanks, >> Niall. >> >> -- >> -- >> >> The information transmitted is intended only for the person or entity to >> which it is addressed and may contain confidential and/or privileged >> material. Statements and opinions expressed in this e-mail may not represent >> those of the sender. Any review, retransmission, dissemination or other use >> of, or taking of any action in reliance upon, this information by persons or >> entities other than the intended recipient is prohibited. If you received >> this in error, please contact the sender immediately and delete the material >> from any computer. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/725829c6/attachment-0002.html From yehavi.bourvine at gmail.com Sun Dec 13 08:05:17 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 18:05:17 +0200 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: I would like all phones have the same general configuration... If no other way, then I'll do that. Thanks, __Yehavi: 2009/12/13 Seven Du > you can use the same ip with different port > > 2009/12/13, Yehavi Bourvine : > > Hello, > > > > In the WIKI page that talks about Freeswitch performance there is a > > sentence: > > > > *libsofia only handles 1 thread per profile, so if that is your bottle > neck > > use more profiles* > > > > How can I enable more than one profile on the same interface? Won't they > > colide when using the same IP and port? > > > > Thanks! __Yehavi: > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/9c6592df/attachment-0002.html From moises.silva at gmail.com Sun Dec 13 08:19:56 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 13 Dec 2009 11:19:56 -0500 Subject: [Freeswitch-users] sangoma a101 - error compiling Wanpipe drivers with TDM API support on ubuntu 9.10 In-Reply-To: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> References: <10128ef10912130656s4a74556eof5743fdffef7280c@mail.gmail.com> Message-ID: On Sun, Dec 13, 2009 at 9:56 AM, Tzury Bar Yochay wrote: > Hi all, > > I am getting a frustrating errors while trying to build the latest wanpipe > on my newly and freshly installed Ubuntu 9.10 > > I pastebin the output at http://gist.github.com/255442 and would > appreciate any help. > > Hi Tzury, It seems the kernel developers decided to move some members of the net_device kernel structure to an internal structure called net_device_ops netdev_ops, I see Ubuntu 9.10 uses a pretty recent kernel. The engineer in charge of the wanpipe drivers will likely soon put a fix. In the meantime you will have to downgrade the kernel somehow. It is usually recommended to use CentOS for servers running wanpipe, I strongly recommend you to switch if possible. If changing to CentOS is not an option, you will have to wait for the wanpipe drivers to be fixed for this recent kernel. You can always send an e-mail to support at sangoma.com to let them know and that may speed things up. PD. I will answer your other e-mail soon. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/8cff8a7c/attachment-0002.html From abeka at greatiam.com Sun Dec 13 08:19:05 2009 From: abeka at greatiam.com (Otis) Date: Sun, 13 Dec 2009 16:19:05 +0000 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <20091213152158.GV31924@base.carmickle.com> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> Message-ID: <4B2513F9.4020000@greatiam.com> Frank Carmickle wrote: > On Sun, Dec 13, Otis wrote: > >> Sorry >> >> I posted this earlier but did not do the due diligence and sent it with >> so much typo them meaning does not come out: >> >> In a nutshell I would like to know : >> >> 1. How FS would know which dialplan to use for an extension with user >> context other than default. >> > > It just uses the context tag which you include extensions in side it. > > > > > > > > > > >> 2. If a file file has to be created does the name matter >> > > Doesn't matter. The include statements pull the files in. Wrap the stuff in the included file in tags. > > >> 3. Where should that file be located. >> > > Anywhere! It easiest to set up includes like > > > > Now you can put all kinds of files in the public dir and they will get included when the preprocess runs. The preprocess runs at start up so you need to restart IIRC. > > HTH > --FC > > > Thank you so much for your time. From mrene_lists at avgs.ca Sun Dec 13 08:26:11 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 13 Dec 2009 11:26:11 -0500 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: <57B69B54-9477-4AEB-B16F-DA235C646A48@avgs.ca> You can have multiple SRV records pointing to different ports. See: http://mit.edu/sip/sip.edu/dns.shtml Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Dec-09, at 11:05 AM, Yehavi Bourvine wrote: > I would like all phones have the same general configuration... If no > other way, then I'll do that. > > Thanks, __Yehavi: > > 2009/12/13 Seven Du > you can use the same ip with different port > > 2009/12/13, Yehavi Bourvine : > > Hello, > > > > In the WIKI page that talks about Freeswitch performance there > is a > > sentence: > > > > *libsofia only handles 1 thread per profile, so if that is your > bottle neck > > use more profiles* > > > > How can I enable more than one profile on the same interface? > Won't they > > colide when using the same IP and port? > > > > Thanks! __Yehavi: > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/d1db0855/attachment-0002.html From frank at carmickle.com Sun Dec 13 08:29:32 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 11:29:32 -0500 Subject: [Freeswitch-users] Sofia performance In-Reply-To: References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> Message-ID: <20091213162932.GW31924@base.carmickle.com> On Sun, Dec 13, Yehavi Bourvine wrote: > I would like all phones have the same general configuration... If no other > way, then I'll do that. Have you already set up a system and found the load of all your phones to be to high? How many phones are we talking about? A load balancer is a solution if you've already tweaked the system for maximum performance. --FC From frank at carmickle.com Sun Dec 13 08:35:56 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sun, 13 Dec 2009 11:35:56 -0500 Subject: [Freeswitch-users] Link between Use-context and dialplan In-Reply-To: <115561884BFA43AAA5CACA534551E5F1@cune.pri> References: <4B246218.2020804@greatiam.com> <20091213152158.GV31924@base.carmickle.com> <115561884BFA43AAA5CACA534551E5F1@cune.pri> Message-ID: <20091213163556.GX31924@base.carmickle.com> On Sun, Dec 13, Russell Mosemann wrote: > Frank Carmickle wrote: > > Now you can put all kinds of files in the public dir and they will get > > included when the preprocess runs. The preprocess runs at start up so you > > need to restart IIRC. > > Or fire up /bin/fs_cli and issue a "reloadxml" or "reload ", if a change affects a module. Thanks for clearing that up for me. I seemed to remember that the wiki was unclear about preprocessor variables being updated with a reloadxml but it is clear now if it wasn't before. --FC From yehavi.bourvine at gmail.com Sun Dec 13 10:21:45 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 13 Dec 2009 20:21:45 +0200 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <20091213162932.GW31924@base.carmickle.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> Message-ID: We are still on a small "proof of concept" system, but I am looking at the future... Thanks, __Yehavi: 2009/12/13 Frank Carmickle > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all phones have the same general configuration... If no > other > > way, then I'll do that. > > Have you already set up a system and found the load of all your phones to > be to high? How many phones are we talking about? A load balancer is a > solution if you've already tweaked the system for maximum performance. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/3716d911/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 13 10:29:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 12:29:20 -0600 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> Message-ID: <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> The swig java one is almost done we need someone who likes java to finish it but as you can see most java ppl seem to always want to do it "their way" On Dec 13, 2009 9:56 AM, "Niall Crosby" wrote: Pure Java is my preference - am looking to build apps that are portable. N. 2009/12/13 Jo?o Mesquita > > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 PM, Niall Crosby References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> Message-ID: <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> Here is my standard asvice on people worrying about performance before even trying fs. Rule of thumb, if you have ever used asterisk, multiply everything by 10 so if there is a performance concern assume it will not arise unless you get at least 10 asterisks worth of performance first. People do thosands of channels with media and tens of thousands with no media, try it first before freting about imaginary load concerns. On Dec 13, 2009 12:28 PM, "Yehavi Bourvine" wrote: We are still on a small "proof of concept" system, but I am looking at the future... Thanks, __Yehavi: 2009/12/13 Frank Carmickle > > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all phones have the same general config... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/5345bf5f/attachment-0002.html From niall.crosby at gmail.com Sun Dec 13 10:42:56 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Sun, 13 Dec 2009 18:42:56 +0000 Subject: [Freeswitch-users] Java ESL In-Reply-To: <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> Message-ID: <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> Where can I find the Java swig ESL? I like Java so am happy to put time towards generating a pure Java ECL, however I haven't programmed C in 10+ years so feel like swig would be to much in the deep end for me. 2009/12/13 Anthony Minessale > The swig java one is almost done we need someone who likes java to finish > it but as you can see most java ppl seem to always want to do it "their way" > > On Dec 13, 2009 9:56 AM, "Niall Crosby" wrote: > > > Pure Java is my preference - am looking to build apps that are portable. > > N. > > 2009/12/13 Jo?o Mesquita > > > > Can't we just swig it to Java? > > JM > > On Sun, Dec 13, 2009 at 1:05 > PM, Niall Crosby > -- -- The information transmitted is intended only for the person or entity > to which it is add... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the sender. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/72d39367/attachment-0002.html From jaybinks at gmail.com Sun Dec 13 13:57:39 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 14 Dec 2009 07:57:39 +1000 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> Message-ID: <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> I'm interested in what the upper limit would be, when expecting a performance improvement with sofia profiles. For example let's say I were to direct connect to customers ( layer 2 ) with a .1q trunk coming in to fs and a Sofia profile for each customer. Am I going to hit a bottleneck at 20,50,100,500 ??? Guess it's hardware limited , but any thoughts ? J On 14/12/2009, at 4:36, Anthony Minessale wrote: > Here is my standard asvice on people worrying about performance > before even trying fs. > > Rule of thumb, if you have ever used asterisk, multiply everything > by 10 so if there is a performance concern assume it will not arise > unless you get at least 10 asterisks worth of performance first. > > People do thosands of channels with media and tens of thousands with > no media, try it first before freting about imaginary load concerns. > >> On Dec 13, 2009 12:28 PM, "Yehavi Bourvine" > > wrote: >> >> We are still on a small "proof of concept" system, but I am >> looking at the future... >> >> Thanks, __Yehavi: >> >> 2009/12/13 Frank Carmickle >> > > On Sun, Dec 13, Yehavi Bourvine wrote: > > I would like all >> phones have the same general config... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/1cc75737/attachment-0002.html From quentusrex at gmail.com Sun Dec 13 14:33:09 2009 From: quentusrex at gmail.com (William King) Date: Sun, 13 Dec 2009 14:33:09 -0800 Subject: [Freeswitch-users] CDR question. Message-ID: <4B256BA5.2050006@gmail.com> Anyone know a good way to determine which extension picked up a call that was bridged to 10+ extensions? -William From anthony.minessale at gmail.com Sun Dec 13 14:37:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 16:37:24 -0600 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> Message-ID: <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> Sep processes does better than sep profiles. We need to push the sofia devs to work on a better concurrancy scheme but they are too busy with other nokia duties these days so were stuck with what we got for now. About 400cps on a good day On Dec 13, 2009 4:05 PM, "Jay Binks" wrote: I'm interested in what the upper limit would be, when expecting a performance improvement with sofia profiles. For example let's say I were to direct connect to customers ( layer 2 ) with a .1q trunk coming in to fs and a Sofia profile for each customer. Am I going to hit a bottleneck at 20,50,100,500 ??? Guess it's hardware limited , but any thoughts ? J On 14/12/2009, at 4:36, Anthony Minessale wrote: > Here is my standa... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/2eb5b1b8/attachment-0002.html From anthony.minessale at gmail.com Sun Dec 13 14:38:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 13 Dec 2009 16:38:41 -0600 Subject: [Freeswitch-users] Java ESL In-Reply-To: <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> References: <4aec92830912130705p3b0ec966h3ecab54d9a01ef9b@mail.gmail.com> <4aec92830912130750h7aada848xa05f7e928d3f58fc@mail.gmail.com> <191c3a030912131029s757a7b20o484b1df05f2cd2bc@mail.gmail.com> <4aec92830912131042k563715a6x2a624df2f219bf9f@mail.gmail.com> Message-ID: <191c3a030912131438u56863e73h3f95f38b0e536848@mail.gmail.com> In the libs/esl there is already swigged java but I don't know how to load it etc try make javamod in esl On Dec 13, 2009 12:47 PM, "Niall Crosby" wrote: Where can I find the Java swig ESL? I like Java so am happy to put time towards generating a pure Java ECL, however I haven't programmed C in 10+ years so feel like swig would be to much in the deep end for me. 2009/12/13 Anthony Minessale > > The swig java one is almost done we need someone who likes java to finish it but as you can see... -- -- The information transmitted is intended only for the person or entity to which it is addressed and ... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/74566495/attachment-0002.html From dfansler at dv-fansler.com Sun Dec 13 17:27:39 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Sun, 13 Dec 2009 20:27:39 -0500 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> Message-ID: <02cd01ca7c5c$a05b64a0$e1122de0$@com> Final Count was just over 900 files. At the moment I am putting them in a logical order - as best I can tell with my limited experience - combining chapters and providing links from the table of contents to sections in each chapter. Then I will go back in retarget all the hyperlinks to point to the document rather than the wiki site. This may take a few hours J David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert Clayton Sent: Saturday, December 12, 2009 10:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch Documentation David, Thanks for your hard work. Maybe more organization will make the areas needing substance or explanation more obvious. Bob On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler wrote: I am new to FreeSWITCH (ok a month old) and am still learning as hard as I can. In the recent talk about documentation, I had noticed that finding documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to download all the wiki pages and put them in some sort of logical order. No I am not finished yet, but does anyone else out there realize how many wiki pages there are on FreeSWITCH's site? So far I am up to 767 pages and have another 100 to go. I had no idea what I was getting into! David David V. Fansler s/v Annabelle dfansler at dv-fansler.com www.dv-fansler.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091213/f708e9cc/attachment-0002.html From jingwei.yang at gmail.com Sun Dec 13 18:11:38 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 14 Dec 2009 10:11:38 +0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: <13529f9d0912131811r5ea76df9g37d961ff949dd5e5@mail.gmail.com> Thanks David. It'll be of a great help to FS newbies like me. Regards, -Jingwei On Mon, Dec 14, 2009 at 9:27 AM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Robert > Clayton > *Sent:* Saturday, December 12, 2009 10:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch Documentation > > > > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler < > dfansler at dv-fansler.com> wrote: > > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/89dcd08f/attachment-0002.html From dome at tel.co.th Sun Dec 13 18:41:57 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 14 Dec 2009 09:41:57 +0700 Subject: [Freeswitch-users] Event Socket outbound in PHP Message-ID: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> Dear All, Now i use php for ESL outbound. i get variable from stdin and process. (i use xinetd for handle socket) $in = fopen("php://stdin", "r"); Problem is when i use read command for get input from DTMF. i can't get variable. So now i use 2 php script. and use read appliction in XML DIalplan for solve this problem. I plan to use php handle socket like a perl in http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl But i want to know how PHP work like this example ? my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); Can someoue help me ? Best Regards. Dome C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b5d489b5/attachment-0002.html From thangappan143 at gmail.com Sun Dec 13 21:10:28 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 14 Dec 2009 10:40:28 +0530 Subject: [Freeswitch-users] Fwd: Getting started on IVR Library In-Reply-To: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> Message-ID: <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> I've seen the source code of mod_rss and mod_voicemail. I was not able to get it. What are the steps do I need to follow for implementing IVR using IVR library? Is there any documentations available for knowing about IVR library? I might be wrong please correct me... Please help me............ ---------- Forwarded message ---------- From: Thangappan.M Date: Sat, Dec 12, 2009 at 1:12 PM Subject: Getting started on IVR Library To: freeswitch-users Dear all , I've seen the IVR library functions which are implemented in C language. Can any one please suggest how can I use that library or give idea to do the IVR programs in C through this library. Please help me....... -- Regards, Thangappan.M -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0a841290/attachment-0002.html From info at daccii.it Mon Dec 14 02:39:50 2009 From: info at daccii.it (Albano Daniele Salvatore - Lavoro) Date: Mon, 14 Dec 2009 11:39:50 +0100 Subject: [Freeswitch-users] embedded freeswitch compatable hardware In-Reply-To: <20091211024228.GE14547@sys11.in.vpac.org> References: <20091210005545.GD28041@sys11.in.vpac.org> <2d9149cd0912091947x801ec6dv5969b7c6bf7c9e71@mail.gmail.com> <4B207ECC.1020405@microcomaustralia.com.au> <20091211015724.GC14547@sys11.in.vpac.org> <20091211024228.GE14547@sys11.in.vpac.org> Message-ID: <4B2615F6.7080701@daccii.it> Hi, you could build an "embedded filesystem" using one of these great tools: - buildroot crosstools-ng - ptxdist - moblin2 - clfs The one i prefer is buildroot because is simply, flexible and fast to build a root filesystem: to setup a minimal system, once you builded the toolchain using crosstools-ng, you need only a couple of hours! It support a lot of hardware platform and br mailing list is VERY active! If you need to use the hardware supplier toolchain you can, just set it instead to build one with crosstools-ng! Freeswitch uses threads a lot, more than asterisk i think, so is preferable to use glibc or eglibc (more configurable than glibc, but essentially the same) instead of uclibc because these libraries are great for embedded hardware but support only old style threads (linuxthread instead of nptl). You can use busybox as base for a lot of stuff and then build what you need (php with fastcgi, lighttpd [althought i prefer nginx], sqlite, lua and so on). I'm doing some job integrating fs with an own software, so i'll start to work on it, probably, the next year, at the end of first quarter or at start of the second, until that moment i'll use a shrinked down ubuntu 8.04 to fit it on a 4gb cf. (i got my alix board a couple of days ago!). However i've used br in a couple of personal project and to help my brother at university (needed to setup a system on a xilinx virtex ii board) If you prefer something more well tested you can give a try to ptxdist or moblin2 but them are harder to use and configure althought them works better! Best Regards, Daniele Brian May ha scritto: > On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote: >> As a note, we are pretty aggressive about making sure all this stuff works >> right out of svn without any patches so it should be easy to port freeswitch >> to most platforms now. > > Thats good to hear. > > I am guessing this means I should use a recent version. I see there is an > Ubuntu archive, wondering if that will work with Voyage Linux. If not, I should > be able to build from the source. > > Anyway I sent an email to Yawarra to ask them if the net5501 computer > is compatible with > the TDM400 cards. There is something about a kit for the dual rack mount > computer for the TDM400, which would be good if I had a rack, and somewhere to > put a rack. So presumably this means it should work for the non-rack mount > system too. -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 381 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/bb88053f/attachment-0002.vcf From Prometheus001 at gmx.net Mon Dec 14 04:05:32 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 14 Dec 2009 13:05:32 +0100 Subject: [Freeswitch-users] Which ATAs to chose for modem connections? Message-ID: <4B262A0C.4070604@gmx.net> We currently use Patton gateways SN4116 for attaching fax and modem equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, so media flow goes the following way: Modem/Fax => Patton_SN4116 => Patton_SN46XX =>PSTN/ISDN However modem connections are not very reliable. We exchanged the SN4118 against a "Fritzbox" ATA and the situation improves. However Fritzboxes do not deliver the number of ports we need. What is your experience? Which ATA is the best choice for modem connections? Best regards Peter From steveu at coppice.org Mon Dec 14 04:29:02 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 14 Dec 2009 20:29:02 +0800 Subject: [Freeswitch-users] Which ATAs to chose for modem connections? In-Reply-To: <4B262A0C.4070604@gmx.net> References: <4B262A0C.4070604@gmx.net> Message-ID: <4B262F8E.7040003@coppice.org> On 12/14/2009 08:05 PM, Peter P GMX wrote: > We currently use Patton gateways SN4116 for attaching fax and modem > equipment to our Freeswitch system. Freeswitch is in bypass-media-mode, > so media flow goes the following way: > Modem/Fax => Patton_SN4116 => Patton_SN46XX =>PSTN/ISDN > However modem connections are not very reliable. We exchanged the SN4118 > against a "Fritzbox" ATA and the situation improves. However Fritzboxes > do not deliver the number of ports we need. > > What is your experience? Which ATA is the best choice for modem connections? > Best regards > Peter > > Your improvement is probably more due to luck than engineering. Turning off any dynamic jitter buffering, and making sure you use only A-law or u-law, will reduce the failures. the only thing that will give reliable results for modems is a managed, absolutely lossless, packet path, and terminal equipment which does doesn't slip data samples to allow for mismatched sampling clocks. Good luck finding those. Steve From oscav at hotmail.fr Mon Dec 14 04:45:19 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 14 Dec 2009 04:45:19 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? Message-ID: <26777181.post@talk.nabble.com> Hi, What are the solutions to support the G729/G723 codec within FreeSwitch ? Thanks -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Mon Dec 14 05:53:57 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 14 Dec 2009 05:53:57 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? Message-ID: <26778101.post@talk.nabble.com> Hello I'm a bit confused at the difference between those two concepts. Contexts are created in the /dialplan, and are refered to by items in /SIP_profiles and extensions in /directory. What purpose do contexts and profiles play? Thank you. -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26778101.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Dec 14 07:19:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:19:56 -0600 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26777181.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> Message-ID: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Software G729 will be available by the end of the month. As for, G723 we are not currently working on it. On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: > > Hi, > > What are the solutions to support the G729/G723 codec within FreeSwitch ? > > Thanks > > > -- > View this message in context: > http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/d28816c5/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 14 07:29:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:29:20 -0600 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <26778101.post@talk.nabble.com> References: <26778101.post@talk.nabble.com> Message-ID: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Profile is a collection of preferences uses by conferences etc. In the case of SIP a profile is also the name for the resulting SIP UA created by a particular profile. Context is a narrowed down view of something, in the case of the dialplan a context is a set of extensions. It's like having a dedicated set of extensions per distinct context name like parallel universes. both the foo context and the bar context can have extension 2001. On Mon, Dec 14, 2009 at 7:53 AM, Fred-145 wrote: > > Hello > > I'm a bit confused at the difference between those two concepts. Contexts > are created in the /dialplan, and are refered to by items in /SIP_profiles > and extensions in /directory. > > What purpose do contexts and profiles play? > > Thank you. > -- > View this message in context: > http://old.nabble.com/Context-vs.-profile--tp26778101p26778101.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b3d17059/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 14 07:37:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 09:37:32 -0600 Subject: [Freeswitch-users] Link between User context and dialplan In-Reply-To: <4B24249A.1040004@greatiam.com> References: <4B24249A.1040004@greatiam.com> Message-ID: <191c3a030912140737ldb2dbe7of181c4b6cfb22f79@mail.gmail.com> a dial plan is a another level of indirection on top of contexts it denotes a specific module which implements the entire universe of dialing with room for as many contexts as you have room for. There is an XML dialplan, an ENUM dialplan etc. You can write your own dialplan and send calls to it. On Sat, Dec 12, 2009 at 5:17 PM, Otis wrote: > Hi folks > > I am so sorry if this is such a basic thing. > > well, when a user/extension eg 8888 is created in with say a user > context - SWAHILI-SPEAKERS Please hear are my questions: > > 1. What dialplan will that user/extn use. > 2. I guess I have to create a dialplan Should the dial-plan also be > called WAHILI-SPEAKERS (is the case relevant ) ? Or could it be > any name ? > 3. And how does FS know to load that dialplan for that user. > 4. Where should that xml file be stored ? > 5. Is there a means of determinig which dialplan was used for a call ? > > Thanks I think I have demonstrated enough thickness for now > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/92a80dfa/attachment-0002.html From helmut.kuper at ewetel.de Mon Dec 14 08:00:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 14 Dec 2009 17:00:03 +0100 Subject: [Freeswitch-users] bidirectional SRTP Message-ID: <4B266103.7060305@ewetel.de> Hello, today I found that my SRTP calls are only SRTP in one way (from snom to FS). The other direction is still RTP (at least if I believe wireshark). How can I get both directions using SRTP? I do a simple call to an announcement in FS. My SIP-profile has: I followed the document of FS Wiki: http://wiki.freeswitch.org/wiki/Secure_RTP But it seems there is no ch_var like sip_has_crypto. Is there a way to have both directions/streams per leg using SRTP? Please find attached the pcap trace of my snom phone. There you can see that snom offers crypto to FS. FS sends an OK with SRTP and RTP. When you look into the RTP data you see SRTP from snom to FS and RTP from FS to snom. Any hints? regards Helmut -------------- next part -------------- A non-text attachment was scrubbed... Name: trace.pcap Type: application/octet-stream Size: 79985 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/323eb17d/attachment-0002.obj From steveu at coppice.org Mon Dec 14 08:02:40 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 15 Dec 2009 00:02:40 +0800 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Message-ID: <4B2661A0.6050208@coppice.org> On 12/14/2009 11:19 PM, Anthony Minessale wrote: > Software G729 will be available by the end of the month. > As for, G723 we are not currently working on it. There is a legit option for G.723.1 - the Digium TC400B card. Its supported by Freeswitch, thanks to Moises. > > > On Mon, Dec 14, 2009 at 6:45 AM, Oscav > wrote: > > > Hi, > > What are the solutions to support the G729/G723 codec within > FreeSwitch ? > Steve From mrene_lists at avgs.ca Mon Dec 14 08:05:47 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 14 Dec 2009 11:05:47 -0500 Subject: [Freeswitch-users] bidirectional SRTP In-Reply-To: <4B266103.7060305@ewetel.de> References: <4B266103.7060305@ewetel.de> Message-ID: <36F785FB-8FCF-436C-BB80-9C9B34F013CF@avgs.ca> Both the INVITE and the 200 OK have an a=crypto line. You need to know that only the rtp payload is encrypted, it is normal that you see the headers as-is. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 14-Dec-09, at 11:00 AM, Helmut Kuper wrote: > Hello, > > > today I found that my SRTP calls are only SRTP in one way (from snom > to FS). The other direction is still RTP (at least if I believe > wireshark). How can I get both directions using SRTP? > > I do a simple call to an announcement in FS. > > My SIP-profile has: > > > I followed the document of FS Wiki: > http://wiki.freeswitch.org/wiki/Secure_RTP > > But it seems there is no ch_var like sip_has_crypto. > > Is there a way to have both directions/streams per leg using SRTP? > > Please find attached the pcap trace of my snom phone. > > There you can see that snom offers crypto to FS. FS sends an OK with > SRTP and RTP. When you look into the RTP data you see SRTP from snom > to FS and RTP from FS to snom. > > > Any hints? > > regards > Helmut > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Mon Dec 14 09:12:48 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 14 Dec 2009 09:12:48 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Message-ID: <26779926.post@talk.nabble.com> Thanks Anthony for the tip. Would you say this is a correct representation of things? "Contexts are a set of extensions in conf/dialplan/ (eg. default, public, etc.) Extensions are configured through files in conf/directory/. Each extension maps to a context (). Profiles refer to contexts in a dialplan (eg. ). Profiles are groups of settings used by different parts of the network, eg. Internal (private LAN), External (Internet-accessible, public LAN), etc. Each profile has a unique IP + port number. " -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26779926.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Dec 14 09:21:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 09:21:05 -0800 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> Message-ID: <87f2f3b90912140921t76149076y505e4dfc5eefdb76@mail.gmail.com> Profile is a collection of preferences uses by conferences etc. In the case of SIP a profile is also the name for the resulting SIP UA created by a particular profile. Context is a narrowed down view of something, in the case of the dialplan a context is a set of extensions. It's like having a dedicated set of extensions per distinct context name like parallel universes. both the foo context and the bar context can have extension 2001. A classic example of this is with the default dialing range of 1000-1019. This range appears in all three contexts that are defined in the default configuration: conf/dialplan/public.xml in "public_extensions" conf/dialplan/features.xml in "please_hold" conf/dialplan/default.xml in "Local_Extension" If a call comes in on the public context and gets routed to 1000 (or 1001, 1002,...1019) then it gets handled by "public_extensions" in public.xml. That extension transfers the call to 1000 in the features context (defined in features.xml) which executes a "please hold while I transfer your call" kind of operation and then transfers the call to 1000 in the default context (which is defined in default.xml). Check out those three files. You'll see some cool stuff! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0de43ca5/attachment-0002.html From msc at freeswitch.org Mon Dec 14 09:57:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 09:57:58 -0800 Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <26779926.post@talk.nabble.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> <26779926.post@talk.nabble.com> Message-ID: <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> On Mon, Dec 14, 2009 at 9:12 AM, Fred-145 wrote: > > Thanks Anthony for the tip. > > Would you say this is a correct representation of things? > > "Contexts are a set of extensions in conf/dialplan/ (eg. default, public, > etc.) > > Extensions are configured through files in conf/directory/. Each extension > maps to a context (). > It would be more correct to say that "users are configured in conf/directory" > > Profiles refer to contexts in a dialplan (eg. ). > Profiles are groups of settings used by different parts of the network, eg. > A better way to say this would be that a "SIP profile defines a SIP *User Agent*." (If you don't know what a SIP user agent is that's okay - just know that a profile listens for connections on a particular IP + Port and also sends outbound traffic via the same IP/port.) A SIP profile routes unauth'd incoming calls to a pre-defined dialplan and context. Users who have auth'd to the SIP profile have a little more control - the user_context can be defined at the user level in the directory. (Normally a user will just use the "default" context but it needn't be that way - you could have a context for two different entities, e.g. two different businesses running on the same FS server.) Are you typing up something for posterity's sake? If so let me know. I'll be happy to proof-read the finished product and offer suggestions. -MC Internal (private LAN), External (Internet-accessible, public LAN), etc. > Each profile has a unique IP + port number. " > -- > View this message in context: > http://old.nabble.com/Context-vs.-profile--tp26778101p26779926.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/c7a3fff1/attachment-0002.html From ken at ksac.com Mon Dec 14 10:45:51 2009 From: ken at ksac.com (Kendall Stauffer) Date: Mon, 14 Dec 2009 10:45:51 -0800 Subject: [Freeswitch-users] monday build Message-ID: HI I tried to build the svn last Friday and it didn't make the sphinx dll, so I thought I would wait for the Monday build (web site says update every Monday). Is there going to be a windows build today, and or is the sphinx dll build problem fixed if I build it myself? Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/3d4fa9dd/attachment-0002.html From brian at freeswitch.org Mon Dec 14 10:50:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 12:50:52 -0600 Subject: [Freeswitch-users] monday build In-Reply-To: References: Message-ID: <727A1FFD-7408-4A4A-8A65-775D3DC34E7C@freeswitch.org> I do pre releases and it'll be up shortly had to fix a couple of bugs. I don't do binary releases for windows you'll have to do that yourself or wait. /b On Dec 14, 2009, at 12:45 PM, Kendall Stauffer wrote: > HI > I tried to build the svn last Friday and it didn?t make the > sphinx dll, so I thought I would wait for the Monday build (web site > says update every Monday). > Is there going to be a windows build today, and or is the > sphinx dll build problem fixed if I build it myself? > > Thanks!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/48946613/attachment-0002.html From lon at kickasspixels.com Mon Dec 14 11:11:39 2009 From: lon at kickasspixels.com (Lon Baker) Date: Mon, 14 Dec 2009 11:11:39 -0800 Subject: [Freeswitch-users] Billing solutions information Message-ID: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> Hey everyone, I am researching billing solutions for Freeswitch and want to consolidate the information with what others have found, then add it to the Wiki. There are seems to be a number of billing solutions by commercial providers, claiming they can integrate with Freeswitch, but nothing concrete explaining how far they go. Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, etc? Mod_nibblebill handles the basics of updating a database table. The A2Billing teams says they are planning on adding support for Freeswitch in a few months. ASTPP.org says they support Freeswitch, but the site hasn't been updated since 2008. If you know about any solutions, links to solutions or any information can you send it to me? I will organize it and add it to the wiki. Thanks! Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/604b4b2c/attachment-0002.html From dftoro at yahoo.com Mon Dec 14 11:13:17 2009 From: dftoro at yahoo.com (Diego Toro) Date: Mon, 14 Dec 2009 11:13:17 -0800 (PST) Subject: [Freeswitch-users] fifo caller id In-Reply-To: <23f91030912130454w21319313i82b21b2633cad2e5@mail.gmail.com> Message-ID: <27104.38859.qm@web33501.mail.mud.yahoo.com> Thanks for your answer, I want to resolver the caller id to dialplan level, I know variables fifo_caller_consumer_import and fifo_consumer_caller_import but isn't clear for me how use it. Diego Toro http://lacarretade.blogspot.com/ --- On Sun, 12/13/09, Seven Du wrote: From: Seven Du Subject: Re: [Freeswitch-users] fifo caller id To: freeswitch-users at lists.freeswitch.org Date: Sunday, December 13, 2009, 7:54 AM I think if you listen to CUSTOM FIFO::INFO you can get Caller-Caller-ID-Number on event socket. 2009/12/13 Diego Toro Hello, ? I want to know how can I get caller id after call is out queue fifo, I read about fifo_caller_consumer_import and? fifo_consumer_caller_import variables, but i don't know use it. ? I appreciate any suggestion Diego Toro http://lacarretade.blogspot.com/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0774cb53/attachment-0002.html From jeff at jefflenk.com Mon Dec 14 11:25:03 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 14 Dec 2009 11:25:03 -0800 (PST) Subject: [Freeswitch-users] monday build In-Reply-To: References: Message-ID: <1260818703016-4166167.post@n2.nabble.com> Please post back to the list if you have problems with the windows build! Everything is working as far as I know. If you have an existing build you should delete the following directories and let the scripts download it again. libs\pocketsphinx-0.5.99 <- delete libs\sphinxbase-0.4.99 <- delete Kendall Stauffer wrote: > > HI > I tried to build the svn last Friday and it didn't make the sphinx dll, > so I thought I would wait for the Monday build (web site says update every > Monday). > Is there going to be a windows build today, and or is the sphinx dll > build problem fixed if I build it myself? > > Thanks!! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/monday-build-tp4166045p4166167.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Dec 14 11:45:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 11:45:12 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5pre9 is now available Message-ID: <87f2f3b90912141145t6c193b52yf464f208db7b5d59@mail.gmail.com> FYI, The latest pre-release is now available. Usual information is available here: http://www.freeswitch.org/node/222 Please update as soon as you can. (SVN trunk users do the "make current" thing please.) We need your testing and feedback please! Many thanks for continuing to support FreeSWITCH. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/34da2dd4/attachment-0002.html From dmitry.bely at gmail.com Mon Dec 14 11:47:08 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Mon, 14 Dec 2009 22:47:08 +0300 Subject: [Freeswitch-users] Language settings for demo IVR Message-ID: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> I'm playing with demo IVR from FreeSwitch distribution and have a problem with language settings. I would like to use Russian as a default language for voice messages so I set in vars.xml and installed Russian sound files. It works almost correctly: all phrases are played in Russian, but not explicitly specified .wav files; say for Hello everyone, I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like to connect to the event socket and run some api commands and return them (as opposed to checking SIP, for example). I haven't found anything and I've started to write one in perl using ESL. I'm sure whatever I come up with is going to be pretty ugly and I'd much rather use something else. Has this been done? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From brian at freeswitch.org Mon Dec 14 11:50:45 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 13:50:45 -0600 Subject: [Freeswitch-users] monday build In-Reply-To: <1260818703016-4166167.post@n2.nabble.com> References: <1260818703016-4166167.post@n2.nabble.com> Message-ID: Also Pre9 is up now. /b On Dec 14, 2009, at 1:25 PM, Jeff Lenk wrote: > > Please post back to the list if you have problems with the windows > build! > Everything is working as far as I know. > If you have an existing build you should delete the following > directories > and let the scripts download it again. > > libs\pocketsphinx-0.5.99 <- delete > libs\sphinxbase-0.4.99 <- delete From msc at freeswitch.org Mon Dec 14 12:02:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:02:58 -0800 Subject: [Freeswitch-users] FreeSWITCH Nagios Plugin In-Reply-To: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com> References: <2d9149cd0912141149o4b72cc07pdcf5224f17ae260d@mail.gmail.com> Message-ID: <87f2f3b90912141202n20c54caet6d0eb8d7b0af7601@mail.gmail.com> On Mon, Dec 14, 2009 at 11:49 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like > to connect to the event socket and run some api commands and return > them (as opposed to checking SIP, for example). I haven't found > anything and I've started to write one in perl using ESL. I'm sure > whatever I come up with is going to be pretty ugly and I'd much rather > use something else. Has this been done? > > Thanks! > Kristian, I started something similar a few months back before getting side-tracked. It's the skeleton for a nagios plugin, also written in Perl. Email me off list if you want to have a look... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/b88d918b/attachment-0002.html From msc at freeswitch.org Mon Dec 14 12:20:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:20:10 -0800 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: <87f2f3b90912141220w12083098v8b71559c0640a936@mail.gmail.com> On Sun, Dec 13, 2009 at 5:27 PM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > Thanks for this hard work. Please contact me if/when you need another pair of eyes on this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/0389c3be/attachment-0002.html From msc at freeswitch.org Mon Dec 14 12:29:27 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:29:27 -0800 Subject: [Freeswitch-users] Fwd: Getting started on IVR Library In-Reply-To: <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> References: <7aa29e790912112342g1e4dfcfdkee9975319d76b928@mail.gmail.com> <7aa29e790912132110l332f4445k1563af555c77aec3@mail.gmail.com> Message-ID: <87f2f3b90912141229n3c8edb18r4f3054732759e988@mail.gmail.com> On Sun, Dec 13, 2009 at 9:10 PM, Thangappan.M wrote: > > I've seen the source code of mod_rss and mod_voicemail. I was not able to > get it. > > What are the steps do I need to follow for implementing IVR using IVR > library? > Is there any documentations available for knowing about IVR library? > > I might be wrong please correct me... > > Please help me............ > > What kind of application are you building that requires the use of C? The reason I ask is that the higher level scripting languages and the XML configs allow you to create IVRs with much less coding. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/063d90f0/attachment-0002.html From msc at freeswitch.org Mon Dec 14 12:58:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 12:58:55 -0800 Subject: [Freeswitch-users] IVR apps in lua In-Reply-To: References: Message-ID: <87f2f3b90912141258v7c66abaald4282fd33d7b7705@mail.gmail.com> On Fri, Dec 4, 2009 at 7:58 AM, Neil Patel wrote: > Hi All, > > I haven't found a substantial example of IVR applications implemented in > lua. Can anyone suggest where to look? My issue has to do with appropriate > coding style. > > I am implementing a voice message board application in lua. I want to allow > the user to dial buttons to navigate forward and back in the list of > messages. One way to implement playmessage() is to check for a forward/back > command while playing the current message, and if a command is given to > invoke playmessage() with the prev/next message in the list. However, this > leaves a chain of unreturned playmessage calls on the execution stack (a > recursive function). > > Alternatively, the playmessage() function can return control to its caller > (perhaps a while loop that spins forever) and pass back a code to indicate > the command. The caller acts accordingly. This is non-recursive, but for > anything but simple applications this style becomes tedious as you start > needing to pass back more info and up longer chains of functions. > > Any guidance on this would be appreciated. > > Thanks, > Neil > Sorry for the late response. Tony and I have just gotten to the point in the book we're writing that deals with IVRs so I've started looking at this much more closely. Lua (and Perl, PHP, and the other swig'd langs) support a light OOP way of defining IVR menus (via the IVRMenu class, btw) in a dynamic way. I've got the demo IVR all converted to a Lua script with the exception that I've got a bug to track down when using "menu-sub" as an action. (Says that the menu is invalid when I try it; I am gonna work with bkw when we get a few minutes...) Anyway, here's what I've got so far and I intend to drop it into the scripts/lua directory in the source tree once we confirm that it all works. -- lua_ivr.lua -- -- This script is virtually identical to the demo_ivr defined in conf/autoload_configs/ivr.conf.xml -- It uses the same sound files and mostly the same settings -- It is intended to be used as an example of how you can use Lua to create dynamic IVRs -- -- This hash defines the main IVR menu. It is equivalent to the lines in ivr.conf.xml ivr_def = { ["main"] = undef, ["name"] = "demo_ivr_lua", ["greet_long"] = "phrase:demo_ivr_main_menu", ["greet_short"] = "phrase:demo_ivr_main_menu_short", ["invalid_sound"] = "ivr/ivr-that_was_an_invalid_entry.wav", ["exit_sound"] = "voicemail/vm-goodbye.wav", ["confirm_macro"] = "", ["confirm_key"] = "", ["tts_engine"] = "flite", ["tts_voice"] = "rms", ["confirm_attempts"] = "3", ["inter_digit_timeout"] = "2000", ["digit_len"] = "4", ["timeout"] = "10000", ["max_failures"] = "3", ["max_timeouts"] = "2" } -- top is an object of class IVRMenu -- pass in all 16 args to the constructor to define a new IVRMenu object top = freeswitch.IVRMenu( ivr_def["main"], ivr_def["name"], ivr_def["greet_long"], ivr_def["greet_short"], ivr_def["invalid_sound"], ivr_def["exit_sound"], ivr_def["confirm_macro"], ivr_def["confirm_key"], ivr_def["tts_engine"], ivr_def["tts_voice"], ivr_def["confirm_attempts"], ivr_def["inter_digit_timeout"], ivr_def["digit_len"], ivr_def["timeout"], ivr_def["max_failures"], ivr_def["max_timeouts"] ); -- bindAction args = action, param, digits -- The following bindAction line is the equivalent of this XML from demo_ivr in ivr.conf.xml -- top:bindAction("menu-exec-app", "transfer 9996 XML default", "2"); top:bindAction("menu-exec-app", "transfer 9999 XML default", "3"); top:bindAction("menu-exec-app", "transfer 9991 XML default", "4"); top:bindAction("menu-exec-app", "bridge sofia/${domain}/ 888 at conference.freeswitch.org", "1"); top:bindAction("menu-exec-app", "transfer 1234*256 enum", "5"); top:bindAction("menu-sub", "demo_ivr_submenu","6"); top:bindAction("menu-exec-app", "transfer $1 XML features", "/^(10[01][0-9])$/"); top:bindAction("menu-top", "demo_ivr_lua","9"); -- This hash defines the main IVR sub-menu. It is equivalent to the lines in ivr.conf.xml ivr_sub_def = { ["main"] = undef, ["name"] = "demo_ivr_submenu_lua", ["greet_long"] = "phrase:demo_ivr_sub_menu", ["greet_short"] = "phrase:demo_ivr_main_sub_menu_short", ["invalid_sound"] = "ivr/ivr-that_was_an_invalid_entry.wav", ["exit_sound"] = "voicemail/vm-goodbye.wav", ["confirm_macro"] = "", ["confirm_key"] = "", ["tts_engine"] = "flite", ["tts_voice"] = "rms", ["confirm_attempts"] = "3", ["inter_digit_timeout"] = "2000", ["digit_len"] = "4", ["timeout"] = "15000", ["max_failures"] = "3", ["max_timeouts"] = "2" } -- sub_menu is an object of class IVRMenu -- pass in all 16 args to the constructor to define a new IVRMenu object sub_menu = freeswitch.IVRMenu( ivr_sub_def["main"], ivr_sub_def["name"], ivr_sub_def["greet_long"], ivr_sub_def["greet_short"], ivr_sub_def["invalid_sound"], ivr_sub_def["exit_sound"], ivr_sub_def["confirm_macro"], ivr_sub_def["confirm_key"], ivr_sub_def["tts_engine"], ivr_sub_def["tts_voice"], ivr_sub_def["confirm_attempts"], ivr_sub_def["inter_digit_timeout"], ivr_sub_def["digit_len"], ivr_sub_def["timeout"], ivr_sub_def["max_failures"], ivr_sub_def["max_timeouts"] ); -- Bind the action "menu-top" to the * key sub_menu:bindAction("menu-top","demo_ivr_lua","*"); --sub_menu:execute(session,"demo_ivr_submenu_lua"); -- Run the main menu top:execute(session, "demo_ivr_lua"); -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/958d66a6/attachment-0002.html From msc at freeswitch.org Mon Dec 14 13:15:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Dec 2009 13:15:26 -0800 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> Message-ID: <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: > I'm playing with demo IVR from FreeSwitch distribution and have a > problem with language settings. I would like to use Russian as a > default language for voice messages so I set in vars.xml > > > > and installed Russian sound files. It works almost correctly: all > phrases are played in Russian, but not explicitly specified .wav > files; say for > > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > > I have > > 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File > [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] > [System error : No such file or directory.] > > How to fix this and make it use the correct language? > > What about this in vars.xml? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/2cd240a4/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 14 13:34:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Dec 2009 15:34:48 -0600 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> Message-ID: <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> you need to get the fd number of stdin however you do it in sdp and pass it as the constructor to the esl obj On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: > Dear All, > Now i use php for ESL outbound. i get variable from stdin and > process. (i use xinetd for handle socket) > $in = fopen("php://stdin", "r"); > Problem is when i use read command for get input from DTMF. i > can't get variable. So now i use 2 php script. and use read appliction in > XML DIalplan for solve this problem. > I plan to use php handle socket like a perl in > http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl > But i want to know how PHP work like this example ? > > my $host = $new_sock->sockhost(); > > my $fd = fileno($new_sock); > my $con = new ESL::ESLconnection($fd); > > my $info = $con->getInfo(); > > > Can someoue help me ? > > > Best Regards. > > Dome C. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/d3805515/attachment-0002.html From oscav at hotmail.fr Mon Dec 14 14:01:22 2009 From: oscav at hotmail.fr (Oscav) Date: Mon, 14 Dec 2009 14:01:22 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> Message-ID: <26780206.post@talk.nabble.com> Hi Anthony, What kind of software?? Is there any related licensing cost? Will it be also available for windows ?? Regards, Oscav Anthony Minessale-2 wrote: > > Software G729 will be available by the end of the month. > As for, G723 we are not currently working on it. > > > On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: > >> >> Hi, >> >> What are the solutions to support the G729/G723 codec within FreeSwitch ? >> >> Thanks >> >> >> -- >> View this message in context: >> http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26780206.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dfansler at dv-fansler.com Mon Dec 14 14:13:17 2009 From: dfansler at dv-fansler.com (David V. Fansler) Date: Mon, 14 Dec 2009 14:13:17 -0800 (PST) Subject: [Freeswitch-users] Freeswitch Documentation Message-ID: <29589571.1260828798164.JavaMail.root@whwamui-apprise.pas.sa.earthlink.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/7f1e9248/attachment-0002.html From help at pdscc.com Mon Dec 14 18:01:29 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 14 Dec 2009 18:01:29 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, Message-ID: <20091215020132.1AD1C1DB501@sinclaire.sibble.net> Hmm, I emailed the zfoneproject folks about an hour ago asking about a release date for zfone3 and was surprised about a half hour later with a call from PRZ himself. Here's what I got from the call 1) the currently released version of zfone already has support for secure pbx enrollment 2) the tivi softphone client which I am using on windows mobile and symbian smartphones does not yet have secure pbx enrollment support I contacted Tivi support and the 2.0.7 client with support for secure pbx enrollment is due out close to the end of the year, depending on various factors yada, yada, yada. Am I correct in assuming that connecting via a softphone (eikga) on a windows machine also running the latest official zfone client, and calling 9787 should give me more than just a message saying the following? 1) call is secure 2) welcome to the zrtp enrollment agent 3) thank you for calling (about 1-2 seconds after item 2) then I get a few beeps, I see the zhone client saying security is interrupted and the call drops. This is the same class of behaviour I get with the Tivi clients. On 23 Aug 2009 at 17:09, Brian West wrote: > This is because you didn't install the zrtpagent.lua script and dial > zrtp on your keypad to enroll the FS box as a trusted man in the > middle... which btw will only work with the unreleased zfone3. > > /b > > On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote: > > > I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp > > capable > > endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone > > client, > > 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i > > (symbian > > s60) and an O2 Xda Flame (windows mobile 5). > > > > All 3 endpoints are registered with FS using the default extensions > > of 1000- > > 1003 > > > > With global_setvar zrtp_secure_media=true the zrtp negotiation > > between end > > points happens but the SAS never matches,below is console output for > > a call > > between 2 of the endpoints > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From brian at freeswitch.org Mon Dec 14 18:50:46 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Dec 2009 20:50:46 -0600 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <20091215020132.1AD1C1DB501@sinclaire.sibble.net> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net> Message-ID: if you don't have ZRTP compiled in as per the wiki it won't work... their are a few changes coming to this code soon. /b On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote: > Hmm, I emailed the zfoneproject folks about an hour ago asking about a > release date for zfone3 and was surprised about a half hour later > with a call > from PRZ himself. > > Here's what I got from the call > > 1) the currently released version of zfone already has support for > secure pbx > enrollment > > 2) the tivi softphone client which I am using on windows mobile and > symbian > smartphones does not yet have secure pbx enrollment support > > I contacted Tivi support and the 2.0.7 client with support for > secure pbx > enrollment is due out close to the end of the year, depending on > various > factors yada, yada, yada. > > Am I correct in assuming that connecting via a softphone (eikga) on > a windows > machine also running the latest official zfone client, and calling > 9787 > should give me more than just a message saying the following? > > 1) call is secure > 2) welcome to the zrtp enrollment agent > 3) thank you for calling (about 1-2 seconds after item 2) > > then I get a few beeps, I see the zhone client saying security is > interrupted > and the call drops. > > This is the same class of behaviour I get with the Tivi clients. From lloyd.aloysius at gmail.com Mon Dec 14 19:06:35 2009 From: lloyd.aloysius at gmail.com (Aloysius Thevarajah Lloyd) Date: Mon, 14 Dec 2009 22:06:35 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. Message-ID: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Hi All, I am trying to setup a conference room with pin number authentication. I could not find any wiki documents. If some one help me that would be helpful. Thank you in advance. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091214/1711058e/attachment-0002.html From mrene_lists at avgs.ca Mon Dec 14 19:11:32 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 14 Dec 2009 22:11:32 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. In-Reply-To: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> References: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Message-ID: <877D61E3-4650-4CCD-9E02-FC4F7D523056@avgs.ca> http://wiki.freeswitch.org/wiki/Mod_conference Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 14-Dec-09, at 10:06 PM, Aloysius Thevarajah Lloyd wrote: > Hi All, > > I am trying to setup a conference room with pin number > authentication. I could not find any wiki documents. If some one > help me that would be helpful. > > Thank you in advance. > > > Thanks > Lloyd > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From help at pdscc.com Mon Dec 14 19:38:45 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 14 Dec 2009 19:38:45 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, Message-ID: <20091215033848.1668A1651@sinclaire.sibble.net> I do have it compiled in as per the wiki ;-) That's why I am continually scratching my head on why it's not working. I did have to use the scrooge codec setup otherwise the enrollment messages would play at about 1/10 normal speed. :-( On 14 Dec 2009 at 20:50, Brian West wrote: > if you don't have ZRTP compiled in as per the wiki it won't work... > their are a few changes coming to this code soon. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From nagalenoj at gmail.com Mon Dec 14 20:05:45 2009 From: nagalenoj at gmail.com (Nagalenoj H.) Date: Tue, 15 Dec 2009 09:35:45 +0530 Subject: [Freeswitch-users] Freeswitch Documentation In-Reply-To: <02cd01ca7c5c$a05b64a0$e1122de0$@com> References: <2f7ffb8a0912112308l132fe093rab93f97e9f8a5502@mail.gmail.com> <2f7ffb8a0912121434j50dfd61aq6f91220c1828005f@mail.gmail.com> <01a401ca7ba4$4e449190$eacdb4b0$@com> <02cd01ca7c5c$a05b64a0$e1122de0$@com> Message-ID: Great job. You would hear lots of thanks, if you could hear the entire world.! Thanks. Regards, Nagalenoj H. On Mon, Dec 14, 2009 at 6:57 AM, David V. Fansler wrote: > Final Count was just over 900 files. At the moment I am putting them in > a logical order ? as best I can tell with my limited experience ? combining > chapters and providing links from the table of contents to sections in each > chapter. > > > > Then I will go back in retarget all the hyperlinks to point to the document > rather than the wiki site. This may take a few hours J > > > > David > > > > David V. Fansler > > s/v Annabelle > > dfansler at dv-fansler.com > > www.dv-fansler.com > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Robert > Clayton > *Sent:* Saturday, December 12, 2009 10:42 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch Documentation > > > > David, > > Thanks for your hard work. > Maybe more organization will make the areas needing substance or > explanation more obvious. > > Bob > > On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler < > dfansler at dv-fansler.com> wrote: > > I am new to FreeSWITCH (ok a month old) and am still learning as hard as I > can. In the recent talk about documentation, I had noticed that finding > documentation on the FreeSWITCH wiki was a bit of a chore. So I decided to > download all the wiki pages and put them in some sort of logical order. No > I am not finished yet, but does anyone else out there realize how many wiki > pages there are on FreeSWITCH's site? So far I am up to 767 pages and have > another 100 to go. > > I had no idea what I was getting into! > > David > > David V. Fansler > s/v Annabelle > dfansler at dv-fansler.com > www.dv-fansler.com > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/efbfc88a/attachment-0002.html From dmitry.bely at gmail.com Tue Dec 15 02:58:08 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Tue, 15 Dec 2009 13:58:08 +0300 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> Message-ID: <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote: > > > On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: >> >> I'm playing with demo IVR from FreeSwitch distribution and have a >> problem with language settings. I would like to use Russian as a >> default language for voice messages so I set in vars.xml >> >> ? >> >> and installed Russian sound files. It works almost correctly: all >> phrases are played in Russian, but not explicitly specified .wav >> files; say for >> >> ? ?> ? ? ? ?invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >> >> I have >> >> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File >> >> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] >> [System error : No such file or directory.] >> >> How to fix this and make it use the correct language? >> > What about this in vars.xml? > > data="sound_prefix=$${base_dir}/sounds/en/us/callie"/> Yes, that does the job. Thank you! But it looks a bit inconsistent. Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml. Why duplicate the settings? And another problem is that you cannot easily switch the language for your voice menu. - Dmitry Bely From codecomplete at free.fr Tue Dec 15 03:22:22 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:22:22 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? Message-ID: <26793254.post@talk.nabble.com> Hello I'm about to test Freeswitch on Windows, and would like to make sure I get it right: http://files.freeswitch.org/windows_installer/ contains "freeswitch-1.0.4.exe" from 03-Sep-2009 and "freeswitch.exe" from 07-Dec-2009. Am I correct in understanding that I should run the former, and then manually replace freeswitch.exe with the latest and greatest? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26793254.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 03:22:39 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:22:39 -0800 (PST) Subject: [Freeswitch-users] Context vs. profile? In-Reply-To: <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> References: <26778101.post@talk.nabble.com> <191c3a030912140729x63e6eeafp325f5033ceec6a33@mail.gmail.com> <26779926.post@talk.nabble.com> <87f2f3b90912140957v3a88bdb6leb7cb9484a6fc14@mail.gmail.com> Message-ID: <26793258.post@talk.nabble.com> Are you typing up something for posterity's sake? If so let me know. I'll be happy to proof-read the finished product and offer suggestions. Thanks much for the clarification. I'm used to writing short documentation when I learn a new tool, so 1) it helps me understand how it works, 2) I can perform a new install faster, and 3) it helps newbies get a head-start. So in short: - profiles = User Agents ("end points of a phone call", says Wikipedia) where each profile listens on a given IP/port so that a single Freeswitch server can handle several profiles concurrently, eg. one end-point for internal, authenticated users, and another end-point for incoming calls from a VoIP provider - contexts = tells what a caller can do; a call can go through multiple contexts during the length of the call. In a dialplan, contexts = groups of extensions Thank you. -- View this message in context: http://old.nabble.com/Context-vs.-profile--tp26778101p26793258.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 03:38:11 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 03:38:11 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? In-Reply-To: <26793254.post@talk.nabble.com> References: <26793254.post@talk.nabble.com> Message-ID: <26793434.post@talk.nabble.com> I'll answer my own question ;-) Freeswitch.exe = "svn 15826", ie. a temporary installer until 1.0.5 is available. -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26793434.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 04:32:19 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 04:32:19 -0800 (PST) Subject: [Freeswitch-users] [Windows] Install and update FS.exe? In-Reply-To: <26793434.post@talk.nabble.com> References: <26793254.post@talk.nabble.com> <26793434.post@talk.nabble.com> Message-ID: <26794035.post@talk.nabble.com> I have a suggestion to make for the Windows installer: Make installing FreePBX/FusionPBX an option, as it adds 100MB although not everyone wants a web GUI to manage Freeswitch. My .15? -- View this message in context: http://old.nabble.com/-Windows--Install-and-update-FS.exe--tp26793254p26794035.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ovvenkatesan at gmail.com Tue Dec 15 05:36:42 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 15 Dec 2009 19:06:42 +0530 Subject: [Freeswitch-users] hardware requirement to run voip application Message-ID: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Hi to all, I dont know, whether I can post this question here or not, I dont have any other options . I hope some one will help me here. Here is my question? 1. I have developed my voip application on top of freeSwitch, Its working fine with soft phone. 2. Now, I need to test my application by calling my landline. I am living in India. I did some googling, I got to this hardware, *Linksys SPA3102 *. Is this only hardware enough to run my voip application or need more hardware? I am very new to this platform, and not having knowledge on hardware. Can anyone please suggest me which hardware which is freeswitch friendly? -- Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/637142cb/attachment-0002.html From dome at tel.co.th Tue Dec 15 05:38:13 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 15 Dec 2009 20:38:13 +0700 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> Message-ID: <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> 2009/12/15 Anthony Minessale > you need to get the fd number of stdin however you do it in sdp and pass it > as the constructor to the esl obj > > It's work. Thanks. but i found PHP not good for this case. PHP need more resource. LUA look better. Now i'm testing by mod_lua but i plan to mover LUA work with outbound socket. but not found about lua outbounf socket in WIKI Best Regards. Dome C. > > > On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: > >> Dear All, >> Now i use php for ESL outbound. i get variable from stdin and >> process. (i use xinetd for handle socket) >> $in = fopen("php://stdin", "r"); >> Problem is when i use read command for get input from DTMF. i >> can't get variable. So now i use 2 php script. and use read appliction in >> XML DIalplan for solve this problem. >> I plan to use php handle socket like a perl in >> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl >> But i want to know how PHP work like this example ? >> >> my $host = $new_sock->sockhost(); >> >> my $fd = fileno($new_sock); >> my $con = new ESL::ESLconnection($fd); >> >> my $info = $con->getInfo(); >> >> >> Can someoue help me ? >> >> >> Best Regards. >> >> Dome C. >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/7339352f/attachment-0002.html From stevendt at primrosebank.net Tue Dec 15 06:35:32 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Tue, 15 Dec 2009 14:35:32 -0000 Subject: [Freeswitch-users] hardware requirement to run voip application References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Message-ID: Hi, I am almost as new to all this as you, but, here is the benefit of my "experience" ! I cannot suggest any alternative hardware for you, although I'm sure others will, but can confirm that the SPA3102 is "FreeSwitch Friendly" - or rather, FreeSwitch is SPA3102 Friendly ! I started off with Softphones too, then added a few SIP hardware phones. With FreeSwitch installed and working locally, I then added the SPA3102 to allow me to dial out on the PSTN line. The SPA3102 was fairly straightforward to setup, but see the Wiki for the Linksys bug on RTP packet size that you'll need to adjust in the SPA3102 setup http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo regards Dave ----- Original Message ----- From: ovvenkat To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, December 15, 2009 1:36 PM Subject: [Freeswitch-users] hardware requirement to run voip application Hi to all, I dont know, whether I can post this question here or not, I dont have any other options . I hope some one will help me here. Here is my question? 1. I have developed my voip application on top of freeSwitch, Its working fine with soft phone. 2. Now, I need to test my application by calling my landline. I am living in India. I did some googling, I got to this hardware, Linksys SPA3102 . Is this only hardware enough to run my voip application or need more hardware? I am very new to this platform, and not having knowledge on hardware. Can anyone please suggest me which hardware which is freeswitch friendly? -- Regards Venkatesan OV. ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e1a0c849/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 15 06:49:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Dec 2009 08:49:11 -0600 Subject: [Freeswitch-users] Event Socket outbound in PHP In-Reply-To: <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> References: <8ccbff060912131841j3c23c1eew413b73c7f2633346@mail.gmail.com> <191c3a030912141334g384ec08j1334f46eb9f24a52@mail.gmail.com> <8ccbff060912150538j43b1b8b1t234ccb380d42b1fd@mail.gmail.com> Message-ID: <191c3a030912150649m15e1cbfdkbb11ab67dc48d55e@mail.gmail.com> there is a lua esl wrapper too the are all based on the same C code for the ESL obj. same rule applies get file number of stdin and pass to constructor. I like perl the best for ESL but that's just me. On Tue, Dec 15, 2009 at 7:38 AM, Dome Charoenyost wrote: > > > 2009/12/15 Anthony Minessale > > you need to get the fd number of stdin however you do it in sdp and pass it >> as the constructor to the esl obj >> >> It's work. Thanks. but i found PHP not good for this case. PHP need more > resource. LUA look better. > Now i'm testing by mod_lua but i plan to mover LUA work with outbound > socket. but not found about lua outbounf socket in WIKI > > > Best Regards. > > Dome C. > > > > >> >> >> On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost wrote: >> >>> Dear All, >>> Now i use php for ESL outbound. i get variable from stdin and >>> process. (i use xinetd for handle socket) >>> $in = fopen("php://stdin", "r"); >>> Problem is when i use read command for get input from DTMF. i >>> can't get variable. So now i use 2 php script. and use read appliction in >>> XML DIalplan for solve this problem. >>> I plan to use php handle socket like a perl in >>> http://svn.freeswitch.org/svn/freeswitch/trunk/libs/esl/perl/server2.pl >>> But i want to know how PHP work like this example ? >>> >>> my $host = $new_sock->sockhost(); >>> >>> my $fd = fileno($new_sock); >>> my $con = new ESL::ESLconnection($fd); >>> >>> my $info = $con->getInfo(); >>> >>> >>> Can someoue help me ? >>> >>> >>> Best Regards. >>> >>> Dome C. >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/b9aaedc6/attachment-0002.html From ovvenkatesan at gmail.com Tue Dec 15 07:03:40 2009 From: ovvenkatesan at gmail.com (ovvenkat) Date: Tue, 15 Dec 2009 20:33:40 +0530 Subject: [Freeswitch-users] hardware requirement to run voip application In-Reply-To: References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> Message-ID: <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> Hi Dave, Thank you very much for your quick reply and suggestion. I will try with SPA3102 then. Thanks again. Venkat. On Tue, Dec 15, 2009 at 8:05 PM, Dave Stevenson wrote: > Hi, > > I am almost as new to all this as you, but, here is the benefit of my > "experience" ! > > I cannot suggest any alternative hardware for you, although I'm sure others > will, but can confirm that the SPA3102 is "FreeSwitch Friendly" - or rather, > FreeSwitch is SPA3102 Friendly ! > > I started off with Softphones too, then added a few SIP hardware phones. > > With FreeSwitch installed and working locally, I then added the SPA3102 to > allow me to dial out on the PSTN line. > > The SPA3102 was fairly straightforward to setup, but see the Wiki for the > Linksys bug on RTP packet size that you'll need to adjust in the SPA3102 > setup > > http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo > > regards > Dave > > ----- Original Message ----- > *From:* ovvenkat > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Tuesday, December 15, 2009 1:36 PM > *Subject:* [Freeswitch-users] hardware requirement to run voip application > > Hi to all, > > I dont know, whether I can post this question here or not, I dont have any > other options . I hope some one will help me here. Here is my question? > > 1. I have developed my voip application on top of freeSwitch, Its working > fine with soft phone. > 2. Now, I need to test my application by calling my landline. I am living > in India. > > I did some googling, I got to this hardware, *Linksys SPA3102 *. Is this > only hardware enough to run my voip application or need more hardware? > > I am very new to this platform, and not having knowledge on hardware. Can > anyone please suggest me which hardware which is freeswitch friendly? > > -- > > Regards > Venkatesan OV. > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e08feff6/attachment-0002.html From nicolas at medularis.com Tue Dec 15 08:22:29 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 13:22:29 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? Message-ID: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> I'm looking for the equivalent configuration parameter or option of Asterisk's canreinvite (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is there anything like this for configuring a gateway? (there's no info about it on the wiki). Thanks! Nicolas From kristian.kielhofner at gmail.com Tue Dec 15 08:45:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 15 Dec 2009 11:45:01 -0500 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> Message-ID: <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> Closest thing I've found: http://wiki.freeswitch.org/wiki/Channel_Variables#bypass_media_after_bridge On Tue, Dec 15, 2009 at 11:22 AM, Nicolas Brenner wrote: > I'm looking for the equivalent configuration parameter or option of > Asterisk's canreinvite > (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is > there anything like this for configuring a gateway? (there's no info > about it on the wiki). > > Thanks! > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From oscav at hotmail.fr Tue Dec 15 09:24:19 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 15 Dec 2009 09:24:19 -0800 (PST) Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26798406.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26798406.post@talk.nabble.com> Message-ID: <26798443.post@talk.nabble.com> Howler modules are available only for unix platforms. I'm running FS on windows. PBriffett wrote: > > Howler Technologies have a FS compliant G.729 codec and hardware solution. > Feel free to go and get the free trial > > Anthony Minessale-2 wrote: >> >> Software G729 will be available by the end of the month. >> As for, G723 we are not currently working on it. >> >> >> On Mon, Dec 14, 2009 at 6:45 AM, Oscav wrote: >> >>> >>> Hi, >>> >>> What are the solutions to support the G729/G723 codec within FreeSwitch >>> ? >>> >>> Thanks >>> >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26777181.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p26798443.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From oscav at hotmail.fr Tue Dec 15 09:27:03 2009 From: oscav at hotmail.fr (Oscav) Date: Tue, 15 Dec 2009 09:27:03 -0800 (PST) Subject: [Freeswitch-users] stream G729 RTP payload in passthrough Message-ID: <26798489.post@talk.nabble.com> Hi, Would it be possible to "play" a file that is a RTP payload saved from Wireshark, in order to use the G729 passthrough while playing files to caller?? Thanks -- View this message in context: http://old.nabble.com/stream-G729-RTP-payload-in-passthrough-tp26798489p26798489.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Tue Dec 15 09:54:39 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 15 Dec 2009 09:54:39 -0800 Subject: [Freeswitch-users] One-way Video Message-ID: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> I am trying to bring up a video call, but not having much luck. We are only getting one-way video (i.e. the caller sees far-end video, but the callee does not). I added the H263/H264 tags to the pre-process "global_codec_prefs" and "outbound_codec_prefs" tags in vars.xml. Anyone have hints on making two-way video to work? Best Regards, Jerry From nicolas at medularis.com Tue Dec 15 10:00:19 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 15:00:19 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> Message-ID: <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> Thanks, but I would like to keep FS in the media path. What would be the equivalent of an Asterisk sip.conf's canreinvite=no? On Tue, Dec 15, 2009 at 1:45 PM, Kristian Kielhofner wrote: > Closest thing I've found: > > http://wiki.freeswitch.org/wiki/Channel_Variables#bypass_media_after_bridge > From frank at carmickle.com Tue Dec 15 10:12:54 2009 From: frank at carmickle.com (Frank Carmickle) Date: Tue, 15 Dec 2009 13:12:54 -0500 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> Message-ID: <20091215181254.GA31924@base.carmickle.com> On Tue, Dec 15, Nicolas Brenner wrote: > Thanks, but I would like to keep FS in the media path. What would be > the equivalent of an Asterisk sip.conf's canreinvite=no? It's that way by default. Fs wants to listen for events on a channel in the default config. See bind_meta_app. --FC From nicolas at medularis.com Tue Dec 15 10:26:40 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 15 Dec 2009 15:26:40 -0300 Subject: [Freeswitch-users] Equivalent of canreinvite? In-Reply-To: <20091215181254.GA31924@base.carmickle.com> References: <1b46b4e80912150822y549b6717p60a59bf904449b77@mail.gmail.com> <2d9149cd0912150845i79ed939dib87849edd28151bb@mail.gmail.com> <1b46b4e80912151000i384a6ad4t9d6d7f8e85bb0d5b@mail.gmail.com> <20091215181254.GA31924@base.carmickle.com> Message-ID: <1b46b4e80912151026x22c8a1b4xb639cd23e2835ce9@mail.gmail.com> Thanks! On Tue, Dec 15, 2009 at 3:12 PM, Frank Carmickle wrote: > On Tue, Dec 15, Nicolas Brenner wrote: >> Thanks, but I would like to keep FS in the media path. What would be >> the equivalent of an Asterisk sip.conf's canreinvite=no? > > It's that way by default. ?Fs wants to listen for events on a channel in the default config. ?See bind_meta_app. > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jarrod at fed-com.com Mon Dec 14 20:55:44 2009 From: jarrod at fed-com.com (Jarrod Lash) Date: Mon, 14 Dec 2009 23:55:44 -0500 Subject: [Freeswitch-users] conference room with pin number authentication. In-Reply-To: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> References: <8a19bf2e0912141906o2fdb39d2te0d74272ecc911ce@mail.gmail.com> Message-ID: <8c2388d80912142055y675f245aib2d54903a15ac05@mail.gmail.com> Lloyd, I used this sometime ago to setup ours... http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR -- Jarrod Lash, Federated Communications, LLC. www.fed-com.com Office: +1-412-357-2127 Mobile: +1-412-999-0049 Fax: +1-412-545-8368 On Mon, Dec 14, 2009 at 10:06 PM, Aloysius Thevarajah Lloyd wrote: > Hi All, > > I am trying to setup a conference room with pin number authentication. I > could not find any wiki documents. If some one help me that would be > helpful. > > Thank you in advance. > > > Thanks > Lloyd > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From stevesteffler at shaw.ca Tue Dec 15 08:04:37 2009 From: stevesteffler at shaw.ca (Steve Steffler) Date: Tue, 15 Dec 2009 09:04:37 -0700 Subject: [Freeswitch-users] mod_voicemail question Message-ID: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> Hi all, What is the difference between the mod_voicemail "vm_message_ext" parameter and the "file-extension" parameter? I want all my voicemail in .WAV format except for a couple of extensions which need to be in MP3. I'm getting strange results playing with these settings, for example, after logging into the voicemail, it will say "You have 1 new message. First message at ", and then instead of the voicemail message there will be silence and a long pause. Then it will repeat the message count and loop this behavior. During the silence, I seem to be able to press keys to trigger voicemail events, like for example I am allowed to delete the message (although it isn't playing the message to me, and I am instead hearing silence). Any ideas? Steve From kristian.kielhofner at gmail.com Tue Dec 15 11:06:19 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 15 Dec 2009 14:06:19 -0500 Subject: [Freeswitch-users] stream G729 RTP payload in passthrough In-Reply-To: <26798489.post@talk.nabble.com> References: <26798489.post@talk.nabble.com> Message-ID: <2d9149cd0912151106u696e79c2n12dec12051d9f4b5@mail.gmail.com> If your file is G729 on disk you can play it with mod_native_file. No need to deal with RTP, pcaps, etc. On Tue, Dec 15, 2009 at 12:27 PM, Oscav wrote: > > Hi, > > Would it be possible to "play" a file that is a RTP payload saved from > Wireshark, in order to use the G729 passthrough while playing files to > caller?? > > Thanks -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Tue Dec 15 11:09:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 11:09:09 -0800 Subject: [Freeswitch-users] mod_voicemail question In-Reply-To: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> References: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> Message-ID: <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> On Tue, Dec 15, 2009 at 8:04 AM, Steve Steffler wrote: > > Hi all, > > What is the difference between the mod_voicemail "vm_message_ext" parameter > and the "file-extension" parameter? > vm_message_ext is a channel variable: http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext file-extension is a parameter of the voicemail module: http://wiki.freeswitch.org/wiki/Mod_voicemail#file-extension The former sets for a specific user, the latter for mod_voicemail in general. > > I want all my voicemail in .WAV format except for a couple of extensions > which need to be in MP3. > > I'm getting strange results playing with these settings, for example, after > logging into the voicemail, it will say "You have 1 new message. First > message at ", and then instead of the voicemail message there > will be silence and a long pause. Then it will repeat the message count and > loop this behavior. During the silence, I seem to be able to press keys to > trigger voicemail events, like for example I am allowed to delete the > message (although it isn't playing the message to me, and I am instead > hearing silence). > > Any ideas? > Is this perhaps a recording of silence, so that you might actually be listening to a message? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/d5af3742/attachment-0002.html From msc at freeswitch.org Tue Dec 15 11:35:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 11:35:45 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Dec 18 - Need Your Items Message-ID: <87f2f3b90912151135m4677a553k397b36369a924add@mail.gmail.com> Hello friends, Just to let you know, I have posted the FS weekly conf call agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 It's pretty clean at this point so if you've got things that you'd like to discuss with the group then please add your items to the list. If you have items that require attention, like documentation and janitorial items then by all means drop those on the list as well. One thing we do need to discuss is how we will accomplish screen casting. We are going to have presentations and we want as many people as possible to be able to view the screen while listening to the conference. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/81c8634a/attachment-0002.html From malay.thakershi at continuityhealth.com Tue Dec 15 11:56:41 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 15 Dec 2009 13:56:41 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files Message-ID: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> Hello, I create one WAV file that has: Question + Option 1 + Option 2 + Option 3 + . I noticed towards end of the file Cepstral Allison starts chopping and speeding up. So my question text that gets converted to WAV file using swift EXE looks like: Which is the biggest mammal on land? Select one of the following choices.Or press star to skip the question 1 Parrot 2 Elephant 3 T-Rex 4 Blue Whale And my csharp code looks like: pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", "^\\d", ""); What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? Please help. Malay Thakershi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/d5a8ef7f/attachment-0002.html From lists at redbonez.net Tue Dec 15 12:03:11 2009 From: lists at redbonez.net (Adam Ford) Date: Tue, 15 Dec 2009 13:03:11 -0700 Subject: [Freeswitch-users] Caller ID issue Message-ID: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> I have been having a problem with my outgoing caller ID coming through as Private or Restricted using a PRI + Openzap. My provider claims that it must be a configuration on my end. Is there something I might be missing? Setup is essential the default FreeSWITCH configuration. I realize this is pretty vague, but I don't really know where to start to troubleshoot this issue on my end. Caller ID works from SIP UA to SIP UA as well as for all incoming calls. I don't even know where to configure FreeSWITCH to mask my caller ID as private or restricted. Thank-AF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/f4aea3b1/attachment-0002.html From bcxml at hotmail.com Tue Dec 15 12:11:37 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 12:11:37 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 Message-ID: <26801000.post@talk.nabble.com> I have Freeswitch and Microsoft Speech Server 2007 on the same box When Speech Server initiates a call, I get a sip error message 480 Here is the internal profile trace... freeswitch at HD-T2253CN> freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011: ------------------------------------------------------------------------ INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 FROM: ;epid=55D003BB53;tag=25bf 436a29 TO: CSEQ: 2 INVITE CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTACT: ;automata CONTENT-LENGTH: 340 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 209.172.55.154 s=Microsoft Speech Server session c=IN IP4 209.172.55.154 t=0 0 m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: ;epid=55D003BB53;tag=25bf 436a29 TO: CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Content-Length: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel sofia/inter nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing 12482578002- >19059183027 in context public 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /internal/12482578002 at 127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: ;epid=55D003BB53;tag=25bf 436a29 To: ;tag=gr4aF6aS8tZ0j CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "19059183027" ------------------------------------------------------------------------ recv 383 bytes from tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ ACK sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 FROM: ;tag=25bf436a29;epid=55D0 03BB53 TO: ;tag=gr4aF6aS8tZ0j CSEQ: 2 ACK CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTENT-LENGTH: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/ internal/12482578002 at 127.0.0.1:5080) Ended 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1156 Close Channel sof ia/internal/12482578002 at 127.0.0.1:5080 [CS_DESTROY] Can anyone point me in the right direction ? Thanks Brian -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26801000.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lists at redbonez.net Tue Dec 15 12:50:54 2009 From: lists at redbonez.net (Adam Ford) Date: Tue, 15 Dec 2009 13:50:54 -0700 Subject: [Freeswitch-users] Caller ID issue In-Reply-To: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> References: <009201ca7dc1$a23ce3a0$e6b6aae0$@net> Message-ID: <00a601ca7dc8$4cfd99a0$e6f8cce0$@net> Scratch that, I had my Openzap configured for national, not NI2. Thanks, -AF From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Ford Sent: Tuesday, December 15, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Caller ID issue I have been having a problem with my outgoing caller ID coming through as Private or Restricted using a PRI + Openzap. My provider claims that it must be a configuration on my end. Is there something I might be missing? Setup is essential the default FreeSWITCH configuration. I realize this is pretty vague, but I don't really know where to start to troubleshoot this issue on my end. Caller ID works from SIP UA to SIP UA as well as for all incoming calls. I don't even know where to configure FreeSWITCH to mask my caller ID as private or restricted. Thank-AF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a16f1f7c/attachment-0002.html From msc at freeswitch.org Tue Dec 15 13:27:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 13:27:12 -0800 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26801000.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> Message-ID: <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> On Tue, Dec 15, 2009 at 12:11 PM, bcxml wrote: > > I have Freeswitch and Microsoft Speech Server 2007 on the same box > > When Speech Server initiates a call, I get a sip error message 480 > > Here is the internal profile trace... > > freeswitch at HD-T2253CN> > > freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at > 20:04:05 > .445011: > ------------------------------------------------------------------------ > INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 > FROM: > ;epid=55D003BB53;tag=25bf > 436a29 > TO: > CSEQ: 2 INVITE > CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe > MAX-FORWARDS: 70 > VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 > CONTACT: > 704290e5b4e03b>;automata > CONTENT-LENGTH: 340 > USER-AGENT: RTCC/3.0.0.0 > CONTENT-TYPE: application/sdp > ALLOW: UPDATE > ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify > > v=0 > o=- 0 0 IN IP4 209.172.55.154 > s=Microsoft Speech Server session > c=IN IP4 209.172.55.154 > t=0 0 > m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 > a=rtpmap:114 x-msrta/16000 > a=fmtp:114 bitrate=29000 > a=rtpmap:115 x-msrta/8000 > a=fmtp:115 bitrate=11800 > a=rtpmap:97 RED/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 > FROM: > ;epid=55D003BB53;tag=25bf > 436a29 > TO: > CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe > CSEQ: 2 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel > sofia/inter > nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] > 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing > 12482578002- > >19059183027 in context public > Are you handling "19059183027" in the public context? If so, what is that extension doing with the call? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e6e20ee3/attachment-0002.html From mike at jerris.com Tue Dec 15 13:40:14 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 16:40:14 -0500 Subject: [Freeswitch-users] Language settings for demo IVR In-Reply-To: <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> References: <90823c940912141147t5a3683edr1fc56402b0ef946d@mail.gmail.com> <87f2f3b90912141315w360305fas3b4dfe6ea5515ebb@mail.gmail.com> <90823c940912150258v2bb1f006te7737c231ec14138@mail.gmail.com> Message-ID: <1B812EFC-3A6F-428E-BFEB-A2228CC1A3F8@jerris.com> The issue is the demo ivr does not use phrase macros. The line in ru.xml is for the phrase macros. We should probably change this in the future. Mike On Dec 15, 2009, at 5:58 AM, Dmitry Bely wrote: > On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins wrote: >> >> >> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely wrote: >>> >>> I'm playing with demo IVR from FreeSwitch distribution and have a >>> problem with language settings. I would like to use Russian as a >>> default language for voice messages so I set in vars.xml >>> >>> >>> >>> and installed Russian sound files. It works almost correctly: all >>> phrases are played in Russian, but not explicitly specified .wav >>> files; say for >>> >>> >> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >>> >>> I have >>> >>> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File >>> >>> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav] >>> [System error : No such file or directory.] >>> >>> How to fix this and make it use the correct language? >>> >> What about this in vars.xml? >> >> > data="sound_prefix=$${base_dir}/sounds/en/us/callie"/> > > Yes, that does the job. Thank you! But it looks a bit inconsistent. > Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml. > Why duplicate the settings? And another problem is that you cannot > easily switch the language for your voice menu. > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bcxml at hotmail.com Tue Dec 15 13:47:39 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 13:47:39 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151327i165ed521r650cd70db2c4953e@mail.gmail.com> Message-ID: <26802352.post@talk.nabble.com> I have the following setup.... conf\dialplan\public\VoipMs.xml conf\sip_profiles\external\VoipMs.xml mercutioviz wrote: > > On Tue, Dec 15, 2009 at 12:11 PM, bcxml wrote: > >> >> I have Freeswitch and Microsoft Speech Server 2007 on the same box >> >> When Speech Server initiates a call, I get a sip error message 480 >> >> Here is the internal profile trace... >> >> freeswitch at HD-T2253CN> >> >> freeswitch at HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at >> 20:04:05 >> .445011: >> >> ------------------------------------------------------------------------ >> INVITE sip:19059183027 at 219.175.50.104:5060;transport=tcp SIP/2.0 >> FROM: >> ;epid=55D003BB53;tag=25bf >> 436a29 >> TO: >> CSEQ: 2 INVITE >> CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe >> MAX-FORWARDS: 70 >> VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 >> CONTACT: >> > 704290e5b4e03b>;automata >> CONTENT-LENGTH: 340 >> USER-AGENT: RTCC/3.0.0.0 >> CONTENT-TYPE: application/sdp >> ALLOW: UPDATE >> ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify >> >> v=0 >> o=- 0 0 IN IP4 209.172.55.154 >> s=Microsoft Speech Server session >> c=IN IP4 209.172.55.154 >> t=0 0 >> m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 >> a=rtpmap:114 x-msrta/16000 >> a=fmtp:114 bitrate=29000 >> a=rtpmap:115 x-msrta/8000 >> a=fmtp:115 bitrate=11800 >> a=rtpmap:97 RED/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 >> FROM: >> ;epid=55D003BB53;tag=25bf >> 436a29 >> TO: >> CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe >> CSEQ: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel >> sofia/inter >> nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] >> 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing >> 12482578002- >> >19059183027 in context public >> > > Are you handling "19059183027" in the public context? If so, what is that > extension doing with the call? > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26802352.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Dec 15 14:01:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:01:13 -0500 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <26780206.post@talk.nabble.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> Message-ID: <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> We have not published costs yet, but expect it to be inline with other similar offerings. I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives but we have not yet tested the code on windows. Mike On Dec 14, 2009, at 5:01 PM, Oscav wrote: > > Hi Anthony, > > What kind of software?? Is there any related licensing cost? Will it be also > available for windows ?? From a.alalousi at gmail.com Tue Dec 15 14:00:28 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 15 Dec 2009 22:00:28 +0000 Subject: [Freeswitch-users] REDIRECT 503 not working Message-ID: People, I have a very simple call scenario where calls are hitting FreeSWITCH, and I need to send a 302 REDIRECT to get them to go elsewhere without answering them. It's for a phased migration requirement so that traffic can continue to flow to the current site, but gets redirected to a new site. The old site will eventually be decommissioned. Here is what I have in my conf/dialplan/public/test.xml: FreeSWITCH is sending back the 302 back to the test end-point (eyeBeam 1.5.20 build 54436), but the call is not reaching the specified in the data portion of the redirect application. I know it's sending it because of logs FreeSWITCH end and the info being displayed on eyeBeam's client interface stating Call being forwarded ...etc. ...etc. Has anyone had any similar experiences with a similar setup ? Oh, and one more thing, I have disabled firewalling on both the proxy where eyeBeam is registered and the destination where I'm sending the call. I have also verified that my new destination (also a FreeSWITCH box) is accepting registrations, inviites and able to route calls initiated by eyeBeam when directly registered on it. Has anyone had similar experiences ? better still, has anyone successfully setup FreeSWITCH to be an SBC and can give me feedback ? Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS, CCIE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/224cb6ca/attachment-0002.html From mike at jerris.com Tue Dec 15 14:04:07 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:04:07 -0500 Subject: [Freeswitch-users] One-way Video In-Reply-To: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> References: <3948836C76114214979253B4694358CB@greyhawk.tonecommander.com> Message-ID: <79F2C126-EBF5-403F-BEC0-3B0FB287046E@jerris.com> try just 1 video codec in freeswitch codec prefs and make sure you are using trunk, we fixed quite a few video issues recently. Mike On Dec 15, 2009, at 12:54 PM, Jerry Richards wrote: > I am trying to bring up a video call, but not having much luck. We are only > getting one-way video (i.e. the caller sees far-end video, but the callee > does not). I added the H263/H264 tags to the pre-process > "global_codec_prefs" and "outbound_codec_prefs" tags in vars.xml. > > Anyone have hints on making two-way video to work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/9292db96/attachment-0002.html From mike at jerris.com Tue Dec 15 14:05:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:05:25 -0500 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> References: <004c01ca7dc0$b88865e0$299931a0$@thakershi@continuityhealth.com> Message-ID: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> You can do that with phrase macros. Mike On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: > Hello, I create one WAV file that has: > > Question + Option 1 + Option 2 + Option 3 + ? > > I noticed towards end of the file Cepstral Allison starts chopping and speeding up. > > So my question text that gets converted to WAV file using swift EXE looks like: > > Which is the biggest mammal on land? > Select one of the following choices.Or press star to skip the question > 1 Parrot > 2 Elephant > 3 T-Rex > 4 Blue Whale > > > And my csharp code looks like: > pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", > "^\\d", ""); > > > What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. > > I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? > > Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/bdf3d478/attachment-0002.html From mike at jerris.com Tue Dec 15 14:09:04 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 17:09:04 -0500 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26801000.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> Message-ID: Try turning on debug logs, but from this it looks like its not matching any extensions. Mike On Dec 15, 2009, at 3:11 PM, bcxml wrote: > > I have Freeswitch and Microsoft Speech Server 2007 on the same box > > When Speech Server initiates a call, I get a sip error message 480 > > Here is the internal profile trace... > > 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel > sofia/inter > nal/12482578002 at 127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] > 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing > 12482578002- >> 19059183027 in context public > 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup > sofia > /internal/12482578002 at 127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] > send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: From dule.maillist at gmail.com Tue Dec 15 14:22:53 2009 From: dule.maillist at gmail.com (Dan Le) Date: Tue, 15 Dec 2009 17:22:53 -0500 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> Message-ID: <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string Dan On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: > You can do that with phrase macros. > > Mike > > On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: > > Hello, I create one WAV file that has: > > Question + Option 1 + Option 2 + Option 3 + ? > > I noticed towards end of the file Cepstral Allison starts chopping and > speeding up. > > So my question text that gets converted to WAV file using swift EXE looks > like: > > Which is the biggest mammal on land? > Select one of the following choices. strength='weak'/>Or press star to skip the question > 1 Parrot > 2 Elephant > 3 T-Rex > 4 Blue Whale > > > And my csharp code looks like: > pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, > 5000, "*#", > @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV > ", > > @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", > "^\\d", ""); > > > What happens is, the voice just starts chopping and speeding up between > options. Even though I am not able to say that it only does that towards the > end, I think so. > > I thought, if I break each file into individual WAV instead of 1 big WAV, > it may help? > > Is there a way to play multiple (separate) WAV files in PlayAndGetDigits > function? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/7ec4d037/attachment-0002.html From msc at freeswitch.org Tue Dec 15 14:54:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 14:54:52 -0800 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: References: <26801000.post@talk.nabble.com> Message-ID: <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: > Try turning on debug logs, but from this it looks like its not matching any > extensions. > > Agreed. "console loglevel debug" at the fs cli and then make a test call, capture output, drop into pastebin.freeswitch.org, and post the URL in this thread. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/e759a4ed/attachment-0002.html From brian at freeswitch.org Tue Dec 15 15:02:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Dec 2009 17:02:43 -0600 Subject: [Freeswitch-users] REDIRECT 503 not working In-Reply-To: References: Message-ID: <2D31351D-6577-47DC-B9AC-EB1BE9ACAF6B@freeswitch.org> You have to be careful things like eyebeam will send the invite back to FS1 that did the redirect as if it were the proxy with the request URI as the URI you did in the 302 please post a sip trace of the entire exchange on pastebin. /b On Dec 15, 2009, at 4:00 PM, Ahmed Naji wrote: > People, > > I have a very simple call scenario where calls are hitting > FreeSWITCH, and I need to send a 302 REDIRECT to get them to go > elsewhere without answering them. > > It's for a phased migration requirement so that traffic can continue > to flow to the current site, but gets redirected to a new site. The > old site will eventually be decommissioned. > > Here is what I have in my conf/dialplan/public/test.xml: > > > > > > > > > > FreeSWITCH is sending back the 302 back to the test end-point > (eyeBeam 1.5.20 build 54436), but the call is not reaching the > specified in the data portion of the redirect application. I know > it's sending it because of logs FreeSWITCH end and the info being > displayed on eyeBeam's client interface stating Call being > forwarded ...etc. ...etc. > > Has anyone had any similar experiences with a similar setup ? > > Oh, and one more thing, I have disabled firewalling on both the > proxy where eyeBeam is registered and the destination where I'm > sending the call. I have also verified that my new destination (also > a FreeSWITCH box) is accepting registrations, inviites and able to > route calls initiated by eyeBeam when directly registered on it. > > Has anyone had similar experiences ? better still, has anyone > successfully setup FreeSWITCH to be an SBC and can give me feedback ? > > Regards, > > Ahmed. > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS, CCIE > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From msc at freeswitch.org Tue Dec 15 15:05:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Dec 2009 15:05:21 -0800 Subject: [Freeswitch-users] REDIRECT 503 not working In-Reply-To: References: Message-ID: <87f2f3b90912151505q711c34d3qda112279379a9843@mail.gmail.com> Can you turn on debug and sip trace and pastebin the console output? Reply to this thread with the pastebin URL... I'm sure some of the networking gurus can help. -MC On Tue, Dec 15, 2009 at 2:00 PM, Ahmed Naji wrote: > People, > > I have a very simple call scenario where calls are hitting FreeSWITCH, and > I need to send a 302 REDIRECT to get them to go elsewhere without answering > them. > > It's for a phased migration requirement so that traffic can continue to > flow to the current site, but gets redirected to a new site. The old site > will eventually be decommissioned. > > Here is what I have in my conf/dialplan/public/test.xml: > > > > expression="^(?:7153)(\d+)$"> > data="sip:7153$1 at aaa.bbb.ccc.ddd"/> > > > > > FreeSWITCH is sending back the 302 back to the test end-point (eyeBeam > 1.5.20 build 54436), but the call is not reaching the specified in the data > portion of the redirect application. I know it's sending it because of logs > FreeSWITCH end and the info being displayed on eyeBeam's client interface > stating Call being forwarded ...etc. ...etc. > > Has anyone had any similar experiences with a similar setup ? > > Oh, and one more thing, I have disabled firewalling on both the proxy where > eyeBeam is registered and the destination where I'm sending the call. I have > also verified that my new destination (also a FreeSWITCH box) is accepting > registrations, inviites and able to route calls initiated by eyeBeam when > directly registered on it. > > Has anyone had similar experiences ? better still, has anyone successfully > setup FreeSWITCH to be an SBC and can give me feedback ? > > Regards, > > Ahmed. > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS, CCIE > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/63966943/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 15 15:12:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Dec 2009 17:12:01 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> Message-ID: <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> make sure you are using latest trunk because it should not be sounding choppy. On Tue, Dec 15, 2009 at 4:22 PM, Dan Le wrote: > Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string > > Dan > > On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: > >> You can do that with phrase macros. >> >> Mike >> >> On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: >> >> Hello, I create one WAV file that has: >> >> Question + Option 1 + Option 2 + Option 3 + ? >> >> I noticed towards end of the file Cepstral Allison starts chopping and >> speeding up. >> >> So my question text that gets converted to WAV file using swift EXE looks >> like: >> >> Which is the biggest mammal on land? >> Select one of the following choices.> strength='weak'/>Or press star to skip the question >> 1 Parrot >> 2 Elephant >> 3 T-Rex >> 4 Blue Whale >> >> >> And my csharp code looks like: >> pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, >> 5000, "*#", >> @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV >> ", >> >> @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", >> "^\\d", ""); >> >> >> What happens is, the voice just starts chopping and speeding up between >> options. Even though I am not able to say that it only does that towards the >> end, I think so. >> >> I thought, if I break each file into individual WAV instead of 1 big WAV, >> it may help? >> >> Is there a way to play multiple (separate) WAV files in PlayAndGetDigits >> function? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/396e7ea7/attachment-0002.html From malay.thakershi at continuityhealth.com Tue Dec 15 16:26:31 2009 From: malay.thakershi at continuityhealth.com (Malay Thakershi) Date: Tue, 15 Dec 2009 18:26:31 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> Message-ID: <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> Regarding Choppy issue: How do I know what trunk I am using? I had downloaded windows installer from website and installed it. Then I had to insert few DLL files from the build to get it up and running. Is it possible to only get updated files from the latest trunk? Thank you for help. Also thank "Dan" for suggesting mod_file solution for combining files for playback. Malay Thakershi From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, December 15, 2009 5:12 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PlayAndGetDigits multiple WAV files make sure you are using latest trunk because it should not be sounding choppy. On Tue, Dec 15, 2009 at 4:22 PM, Dan Le wrote: Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string Dan On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris wrote: You can do that with phrase macros. Mike On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote: Hello, I create one WAV file that has: Question + Option 1 + Option 2 + Option 3 + . I noticed towards end of the file Cepstral Allison starts chopping and speeding up. So my question text that gets converted to WAV file using swift EXE looks like: Which is the biggest mammal on land? Select one of the following choices.Or press star to skip the question 1 Parrot 2 Elephant 3 T-Rex 4 Blue Whale And my csharp code looks like: pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3, 5000, "*#", @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV ", @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav", "^\\d", ""); What happens is, the voice just starts chopping and speeding up between options. Even though I am not able to say that it only does that towards the end, I think so. I thought, if I break each file into individual WAV instead of 1 big WAV, it may help? Is there a way to play multiple (separate) WAV files in PlayAndGetDigits function? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a937c3e3/attachment-0002.html From brian at freeswitch.org Tue Dec 15 16:41:56 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Dec 2009 18:41:56 -0600 Subject: [Freeswitch-users] PlayAndGetDigits multiple WAV files In-Reply-To: <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> References: <24C90F64-EC60-414E-8935-869ACD6F3C9F@jerris.com> <914fc92a0912151422j51bd2d49hcff535db819b5bff@mail.gmail.com> <191c3a030912151512o22c625f7m1ed760fd168bd7f5@mail.gmail.com> <008301ca7de6$6a590750$3f0b15f0$@thakershi@continuityhealth.com> Message-ID: <4621A1A0-4CCB-4814-8032-8B1CB33C18D3@freeswitch.org> Compile it yourself is the best bet to get the very latests and greatest code. /b On Dec 15, 2009, at 6:26 PM, Malay Thakershi wrote: > Is it possible to only get updated files from the latest trunk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091215/a6e18387/attachment-0002.html From bcxml at hotmail.com Tue Dec 15 18:27:54 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 18:27:54 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> Message-ID: <26805343.post@talk.nabble.com> Here is the link to the debug log http://pastebin.freeswitch.org/11521 Brian mercutioviz wrote: > > On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: > >> Try turning on debug logs, but from this it looks like its not matching >> any >> extensions. >> >> Agreed. "console loglevel debug" at the fs cli and then make a test call, > capture output, drop into pastebin.freeswitch.org, and post the URL in > this > thread. > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Tue Dec 15 18:28:51 2009 From: senakahks at gmail.com (sky1975) Date: Tue, 15 Dec 2009 18:28:51 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill - no hangup and can call without money in database Message-ID: <1260930531708-4173535.post@n2.nabble.com> Dear Sir, I have successfully installed freeSWITCH and it works fine in passthrough mode. I installed nibblebill and it deduct money from the accounts database and it works fine. but I have two problems. 1. Calls can be initiated even though there is a minus value in accounts database 2. Calls doesn't hangup when it goes to minus values. Any answers are greatly appreciated. This is my dialplan: This is the configuration file; -- View this message in context: http://n2.nabble.com/Mod-nibblebill-no-hangup-and-can-call-without-money-in-database-tp4173535p4173535.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue Dec 15 18:46:41 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Dec 2009 21:46:41 -0500 Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <26805343.post@talk.nabble.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> <26805343.post@talk.nabble.com> Message-ID: <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> Yep, there is your issue.. I missed it when you pasted the extension, its a typo in your condition. Dialplan: sofia/internal/12482578002 at 127.0.0.1:5080 Regex (FAIL) [VoipMs] destination_number(19059183027) =~ /expression=/ break=on-false Notice what it is comparing there .. and notice the typo in your condition. Mike On Dec 15, 2009, at 9:27 PM, bcxml wrote: > > > Here is the link to the debug log > > http://pastebin.freeswitch.org/11521 > > > Brian > > > mercutioviz wrote: >> >> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: >> >>> Try turning on debug logs, but from this it looks like its not matching >>> any >>> extensions. >>> >>> Agreed. "console loglevel debug" at the fs cli and then make a test call, >> capture output, drop into pastebin.freeswitch.org, and post the URL in >> this >> thread. >> -MC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bcxml at hotmail.com Tue Dec 15 18:58:00 2009 From: bcxml at hotmail.com (bcxml) Date: Tue, 15 Dec 2009 18:58:00 -0800 (PST) Subject: [Freeswitch-users] SIP Error Message 480 In-Reply-To: <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> References: <26801000.post@talk.nabble.com> <87f2f3b90912151454t5bec4ejaa21d5dc71039eda@mail.gmail.com> <26805343.post@talk.nabble.com> <8B9419AD-E033-4CE7-8BD9-4530C610B889@jerris.com> Message-ID: <26805522.post@talk.nabble.com> Mike..thank you so much... It works fine now Brian Michael Jerris wrote: > > Yep, there is your issue.. I missed it when you pasted the extension, its > a typo in your condition. > > Dialplan: sofia/internal/12482578002 at 127.0.0.1:5080 Regex (FAIL) [VoipMs] > destination_number(19059183027) =~ /expression=/ break=on-false > > Notice what it is comparing there .. > > field="destination_number"expression="expression="^1?(\d{10})$"> > > and notice the typo in your condition. > > Mike > > On Dec 15, 2009, at 9:27 PM, bcxml wrote: > >> >> >> Here is the link to the debug log >> >> http://pastebin.freeswitch.org/11521 >> >> >> Brian >> >> >> mercutioviz wrote: >>> >>> On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris wrote: >>> >>>> Try turning on debug logs, but from this it looks like its not matching >>>> any >>>> extensions. >>>> >>>> Agreed. "console loglevel debug" at the fs cli and then make a test >>>> call, >>> capture output, drop into pastebin.freeswitch.org, and post the URL in >>> this >>> thread. >>> -MC >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805343.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26805522.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Tue Dec 15 20:58:46 2009 From: freeswitch at aastral.net (Bill W) Date: Tue, 15 Dec 2009 23:58:46 -0500 Subject: [Freeswitch-users] ACLs through proxy Message-ID: <4B286906.7040502@aastral.net> Hi All, I have a FreeSWITCH cluster behind an OpenSIPS proxy/load balancer, and I'd like to be able to use the auth-calls feature in my sip profile in conjunction with the parameter in the directory. In addition to running the INVITEs through the load balancer, I also need to run the REGISTERs through the load balancer because some of my endpoints are behind NAT firewalls, and therefore won't accept incoming calls from IPs other than the IP they registered to. INVITEs from the cluster going to registered endpoints are sent back through the proxy, thereby solving the NAT problem. However, having the proxy in the path effectively negates using IP based ACLS. The functionality I require is as follows: 1. Only allow registration if the endpoint IP matches it's own unique acl CIDR (specified in the directory). 2. Only accept INVITEs from endpoints that authenticate AND match the acl CIDR (again, specified in the directory). Does anyone have any recommendations on the best way to get the auth-calls functionality using an IP other than the IP of the last hop? If not, how hard would it be to add a feature to the auth-calls parameter to accept a channel variable from which to obtain the actual endpoint IP? Thanks! Bill From codecomplete at free.fr Tue Dec 15 23:38:34 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 23:38:34 -0800 (PST) Subject: [Freeswitch-users] [Windows] Stable enough for production use? Message-ID: <26807322.post@talk.nabble.com> Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 15 23:51:32 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 15 Dec 2009 23:51:32 -0800 (PST) Subject: [Freeswitch-users] hardware requirement to run voip application In-Reply-To: <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> References: <47d63d920912150536vd0e38e1v85acafe9b6da2d6d@mail.gmail.com> <47d63d920912150703g3d0a2ae1w3122424dd4527e52@mail.gmail.com> Message-ID: <26807453.post@talk.nabble.com> The 3102 has quite a lot of settings you can play with. Here are two useful documents: SPA-3102 Simplified Users Guide Version 1.1a http://www.jmgtechnology.com.au/spa_3102_guide.pdf Linksys ATA Administrator Guide 3.2.pdf http://www.inphonex.com/download/spa8000-ag.pdf -- View this message in context: http://old.nabble.com/hardware-requirement-to-run-voip-application-tp26795029p26807453.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Wed Dec 16 00:27:41 2009 From: senakahks at gmail.com (Senaka Amarakeerthi) Date: Wed, 16 Dec 2009 17:27:41 +0900 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database Message-ID: Dear Sir, I have successfully installed freeSWITCH and it works fine in passthrough mode. I installed nibblebill and it deduct money from the accounts database and it works fine. but I have two problems. 1. Calls can be initiated even though there is a minus value in accounts database 2. Calls doesn't hangup when it goes to minus values. Any answers are greatly appreciated. This is my dialplan: This is the configuration file; From codecomplete at free.fr Wed Dec 16 01:24:23 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 16 Dec 2009 01:24:23 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Message-ID: <26808383.post@talk.nabble.com> Hello Does someone know of a free service on the web that can check whether the UDP ports on my firewall are open after Freeswitch is up and running? ShieldsUp only scans TCP ports. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26808383.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From asobihoudai at yahoo.com Wed Dec 16 01:58:46 2009 From: asobihoudai at yahoo.com (Paul) Date: Wed, 16 Dec 2009 01:58:46 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <26808383.post@talk.nabble.com> References: <26808383.post@talk.nabble.com> Message-ID: <72220.45962.qm@web111310.mail.gq1.yahoo.com> If you have a shell on an external host, nmap will kindly do that for you quickly and without charge. ----- Original Message ---- From: Fred-145 To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 1:24:23 AM Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Hello Does someone know of a free service on the web that can check whether the UDP ports on my firewall are open after Freeswitch is up and running? ShieldsUp only scans TCP ports. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26808383.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Dec 16 04:07:27 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 06:07:27 -0600 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B286906.7040502@aastral.net> References: <4B286906.7040502@aastral.net> Message-ID: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> use "apply-proxy-acl" on the sofia profile. /b On Dec 15, 2009, at 10:58 PM, Bill W wrote: > > However, having the proxy in the path effectively negates using IP > based > ACLS. From juanbackson at gmail.com Wed Dec 16 04:33:26 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 16 Dec 2009 20:33:26 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels Message-ID: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Hi, I am having problem with around 1 % of the channels always get zombilized. What I want to do is to have a background thread that regularly check all the channels that have been in existance for like > 1 hr, and then check to see if there is any RTP coming in and going out. If there is no RTP, then I just hangup that channel. Does anyone know if there is anyway to do that in a freeswitch module? Which API can I use to accomplish this purpose? Alternatively, is there anyway to configure freeswitch so that it will hangup the calls where there is no media in and out for so many seconds? Thanks, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/6857e9b7/attachment-0002.html From costa.zikalala at gmail.com Wed Dec 16 04:37:38 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Wed, 16 Dec 2009 14:37:38 +0200 Subject: [Freeswitch-users] Basic Question on the Internal Profile Message-ID: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> Hi All I understand that to connect to a SIP Provider you have to (amongst other things) define a Gateway on the External Profile. But some gateways may be defined on the Internal Profile. What kind of gateways would these be and what would be their purpose as most gateways are External? Hope my question makes a bit of sense. Thanks CZ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/d81f49b8/attachment-0002.html From Prometheus001 at gmx.net Wed Dec 16 04:47:57 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 16 Dec 2009 13:47:57 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN Message-ID: <4B28D6FD.6010702@gmx.net> Hello, we have the following scenario: A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For the called FS user, call forwarding has been enabled to another PSTN extension (B) . Result: The calling party does not hear any ringing tone. Here an Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact the same Patton Gateway): PSTN(A)====INVITE===>FS PSTN(A)<===TRYING===>FS FS===INVITE==>PSTN(B) FS<==TRYING===PSTN(B) FS<==RINGING==PSTN(B) PSTN(A)<==PROGRESS===FS FS<===OK======PSTN(B) FS====ACK====>PSTN(B) PSTN(A)<===OK========FS PSTN(A)====ACK======>FS I would expect that FS answers RINGING back to PSTN(A). Instead it only answers SESSION PROGRESS. When PSTN(B) answers, they can hear each other, but there was no ringing tone to PSTN(A) before. Are there any hints to overcome this, besides playing early media to PSTN(A)? Best regards Peter From nameer.kazzaz at gmail.com Wed Dec 16 06:01:05 2009 From: nameer.kazzaz at gmail.com (Nameer Kazzaz) Date: Wed, 16 Dec 2009 14:01:05 +0000 Subject: [Freeswitch-users] xml_rpc.conf Message-ID: <4B28E821.20207@gmail.com> Hi all, Can I set xml_rpc server to run on a specific interface I can set the port but not the ip address to bind to. I have a linux server with more then one interface. I don't want to use iptables to block it. Thanks Nameer From brian at freeswitch.org Wed Dec 16 06:52:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 08:52:09 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Message-ID: Why not just set rtp-timeout-sec on the sofia profile and it'll do that for you. Unless something else is going on. /b On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > Hi, > > I am having problem with around 1 % of the channels always get > zombilized. > > What I want to do is to have a background thread that regularly > check all the channels that have been in existance for like > 1 hr, > and then check to see if there is any RTP coming in and going out. > If there is no RTP, then I just hangup that channel. Does anyone > know if there is anyway to do that in a freeswitch module? Which > API can I use to accomplish this purpose? Alternatively, is there > anyway to configure freeswitch so that it will hangup the calls > where there is no media in and out for so many seconds? > > Thanks, > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From neilp at cs.stanford.edu Wed Dec 16 06:56:04 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Wed, 16 Dec 2009 20:26:04 +0530 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? -Neil On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > looks like ogg devel packages are installed but ogg lib is not? > > > On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > > > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > > unknown format. Looking through the log, I found this during startup: > > > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > > Loading module /usr/local/freeswitch/mod/mod_shout.so > > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > > ogg_sync_wrote** > > > > There don't seem to be any compile-time errors, yet I can't seem to > > eliminate this issue. Any help would be appreciated. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/a10b7eca/attachment-0002.html From peter.olsson at visionutveckling.se Wed Dec 16 07:29:28 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 16 Dec 2009 16:29:28 +0100 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <26807322.post@talk.nabble.com> References: <26807322.post@talk.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55540C9EAA@cooper> Hi, We've been running FS in win32 in a "semi-production" environment for some time now (since version 1.0.1 - following the trunk all the time since then). We use it as both a lab environment - to distribute SIP-trunks to different PBX'es, and also for "real" endpoints for some (about 10-15) of our internal users. Over the time there have been a few issues (now many though), but from 1.0.4 and later it's been very stable for our use, and no memory leaks etc. We haven't tried analog PSTN connections though, only SIP, h323 (opal) and some (not much) use of Sangoma E1 connected to one of our PBX'es. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Fred-145 [codecomplete at free.fr] Skickat: den 16 december 2009 08:38 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] [Windows] Stable enough for production use? Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4b2891a332931788118788! From mike at jerris.com Wed Dec 16 07:44:25 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Dec 2009 10:44:25 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. Thanks Mike On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. > > I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? > > -Neil > > On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > looks like ogg devel packages are installed but ogg lib is not? > > > On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: > > > FreeSWITCH seems to be unable to read MP3 files, citing that it's an > > unknown format. Looking through the log, I found this during startup: > > > > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error > > Loading module /usr/local/freeswitch/mod/mod_shout.so > > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: > > ogg_sync_wrote** > > > > There don't seem to be any compile-time errors, yet I can't seem to > > eliminate this issue. Any help would be appreciated. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/db02c3f0/attachment-0002.html From john_platts at hotmail.com Wed Dec 16 07:59:01 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 16 Dec 2009 09:59:01 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial Message-ID: How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ From mike at jerris.com Wed Dec 16 08:00:41 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Dec 2009 11:00:41 -0500 Subject: [Freeswitch-users] xml_rpc.conf In-Reply-To: <4B28E821.20207@gmail.com> References: <4B28E821.20207@gmail.com> Message-ID: On Dec 16, 2009, at 9:01 AM, Nameer Kazzaz wrote: > Hi all, > Can I set xml_rpc server to run on a specific interface I can set > the port but not the ip address to bind to. I have a linux server with > more then one interface. I don't want to use iptables to block it. No, but you always have the option of using iptables to block it. Mike From jpitcher at nuvio.com Wed Dec 16 08:11:05 2009 From: jpitcher at nuvio.com (Jonathan Pitcher) Date: Wed, 16 Dec 2009 08:11:05 -0800 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: Message-ID: John, To do a click to call in FS you need to have some app that connects to the ESL or Event Socket Layer and runs one of the calls diagramed here ... http://wiki.freeswitch.org/wiki/Mod_commands#originate For use with the ESL just prepend api in front of the originate so your call looks something like: $command = 'api originate user at domain &bridge(user at domain)'; As for programs able to do that from a Microsoft Product, That I am not sure of. Jonathan Pitcher On 12/16/09 9:59 AM, "John Platts" wrote: How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/9d850f19/attachment-0002.html From brian at freeswitch.org Wed Dec 16 08:18:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 10:18:28 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: Message-ID: <54F485C1-2FD2-4950-8617-C4C2F20717F9@freeswitch.org> see scripts/perl/call.cgi /b On Dec 16, 2009, at 9:59 AM, John Platts wrote: > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? > > _________________________________________________________________ > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From john_platts at hotmail.com Wed Dec 16 08:24:07 2009 From: john_platts at hotmail.com (John Platts) Date: Wed, 16 Dec 2009 10:24:07 -0600 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: , Message-ID: You've made my day. ________________________________ > From: jpitcher at nuvio.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 16 Dec 2009 08:11:05 -0800 > Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial > > > > > > > > > John, > > > > To do a click to call in FS you need to have some app that connects to the ESL or Event Socket Layer and runs one of the calls diagramed here ... > > > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > > > For use with the ESL just prepend api in front of the originate so your call looks something like: > > > > $command = 'api originate user at domain &bridge(user at domain)'; > > > > As for programs able to do that from a Microsoft Product, That I am not sure of. > > > > Jonathan Pitcher > > > > > > > > On 12/16/09 9:59 AM, "John Platts" wrote: > > > > > > > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > > > Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? > > > > _________________________________________________________________ > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/ From freeswitch at aastral.net Wed Dec 16 08:24:17 2009 From: freeswitch at aastral.net (Bill W) Date: Wed, 16 Dec 2009 11:24:17 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> Message-ID: <4B2909B1.2030103@aastral.net> That's fantastic! FreeSWITCH ROCKS! I'll update the wiki. Thanks, Bill Brian West wrote: > use "apply-proxy-acl" on the sofia profile. > > /b > > On Dec 15, 2009, at 10:58 PM, Bill W wrote: > >> However, having the proxy in the path effectively negates using IP >> based >> ACLS. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 16 09:09:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 09:09:31 -0800 Subject: [Freeswitch-users] Basic Question on the Internal Profile In-Reply-To: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> References: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> Message-ID: <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> On Wed, Dec 16, 2009 at 4:37 AM, Costa Zikalala wrote: > Hi All > > I understand that to connect to a SIP Provider you have to (amongst other > things) define a Gateway on the External Profile. > But some gateways may be defined on the Internal Profile. What kind of > gateways would these be and what would be their purpose as most gateways are > External? > > Hope my question makes a bit of sense. > > This is a good question. First, remember that the profile names "internal" and "external" are just labels that try to give you a basic idea of what you might use them for. The external profile is generally used for outbound registrations, particularly when the FS server is behind NAT. If your FS server isn't behind NAT then you may want to use the internal profile for outbound registrations, although we recommend that your outbound registrations still be in a profile different than your internal profile where local phones are registering. In these cases it's fine just to remove external.xml, make a copy of internal.xml to something like external-nonat.xml, and then edit it so that it uses ports 5080 and 5081. For the record here's what I did for a user just last week: Old: New: Feel free to tinker and remember that if you break something you can just wipe your conf directory and just run "make samples" again and you'll have a brand new set of default configs. Enjoy! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/3c0e7d09/attachment-0002.html From oscav at hotmail.fr Wed Dec 16 09:57:16 2009 From: oscav at hotmail.fr (Oscav) Date: Wed, 16 Dec 2009 09:57:16 -0800 (PST) Subject: [Freeswitch-users] How to set the Session Name on a SDP? Message-ID: <26815554.post@talk.nabble.com> Hi, Is it possible to set (rewrite) the Session Name in the SDP of a 183 progress sent to inbound ? Many thanks -- View this message in context: http://old.nabble.com/How-to-set-the-Session-Name-on-a-SDP--tp26815554p26815554.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ranjtech at gmail.com Wed Dec 16 10:04:54 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 13:04:54 -0500 Subject: [Freeswitch-users] build errors :( Message-ID: <01c901ca7e7a$47a22900$d6e67b00$@com> Hello All, I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: creating freeswitch cc1: warnings being treated as errors libs/esl/fs_cli.c: In function ?complete?: libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? make[2]: *** [fs_cli-fs_cli.o] Error 1 Any ideas? Thanks RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/ffee79de/attachment-0002.html From brian at freeswitch.org Wed Dec 16 10:13:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 12:13:45 -0600 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01c901ca7e7a$47a22900$d6e67b00$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> Message-ID: <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> What SVN rev? /b On Dec 16, 2009, at 12:04 PM, RR wrote: > Hello All, > > I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: > > creating freeswitch > cc1: warnings being treated as errors > libs/esl/fs_cli.c: In function ?complete?: > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? > make[2]: *** [fs_cli-fs_cli.o] Error 1 > > Any ideas? > > Thanks > RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/30b7491e/attachment-0002.html From bruce.mcalister at blueface.ie Wed Dec 16 10:22:51 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Wed, 16 Dec 2009 18:22:51 +0000 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> Message-ID: <4B29257B.10705@blueface.ie> Hi Michael, Michael Jerris wrote: > I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives > Is support for Solaris and/or OpenSolaris x86 planned as well? Thanks Bruce From ranjtech at gmail.com Wed Dec 16 10:27:03 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 13:27:03 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> Message-ID: <01db01ca7e7d$5f14a240$1d3de6c0$@com> Lost that screen that showed me what rev was downloaded but whatever you get after doing a ?svn up? and ?make current?. I had done a rm ?rf in the /usr/src/freeswitch directory and then did svn up. Should I have done svn co instead? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 16, 2009 1:14 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] build errors :( What SVN rev? /b On Dec 16, 2009, at 12:04 PM, RR wrote: Hello All, I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: creating freeswitch cc1: warnings being treated as errors libs/esl/fs_cli.c: In function ?complete?: libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?int? make[2]: *** [fs_cli-fs_cli.o] Error 1 Any ideas? Thanks RR __________ Information from ESET NOD32 Antivirus, version of virus signature database 4694 (20091216) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/59d2c37a/attachment-0002.html From mrene_lists at avgs.ca Wed Dec 16 10:34:56 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 16 Dec 2009 13:34:56 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01c901ca7e7a$47a22900$d6e67b00$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> Message-ID: I fixed that already, update to the latest trunk. 32/64 bit types mismatch. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 16-Dec-09, at 1:04 PM, RR wrote: > Hello All, > > I know you will probably ask me to check out a fresh copy from svn > trunk and all, but I assure you I have done that yet I keep getting > these errors on make: > > creating freeswitch > cc1: warnings being treated as errors > libs/esl/fs_cli.c: In function ?complete?: > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long > int?, but argument 4 has type ?int? > libs/esl/fs_cli.c:440: warning: format ?%ld? expects type ?long > int?, but argument 4 has type ?int? > make[2]: *** [fs_cli-fs_cli.o] Error 1 > > Any ideas? > > Thanks > RR > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4694 (20091216) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/98417287/attachment-0002.html From msc at freeswitch.org Wed Dec 16 10:38:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 10:38:14 -0800 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01db01ca7e7d$5f14a240$1d3de6c0$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> Message-ID: <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> On Wed, Dec 16, 2009 at 10:27 AM, RR wrote: > Lost that screen that showed me what rev was downloaded but whatever you > get after doing a ?svn up? and ?make current?. I had done a rm ?rf in the > /usr/src/freeswitch directory and then did svn up. Should I have done svn > co instead? > Why did you nuke the fs src dir? Just curious. In your case since you deleted everything you're probably better off just starting from scratch with a svn co. Once you're installed then all you need to do is "make current" b/c make current includes an "svn up" among other things. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/5743980a/attachment-0002.html From kristian.kielhofner at gmail.com Wed Dec 16 11:13:38 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:13:38 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? Message-ID: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Hello everyone, Pastebin here: http://pastebin.freeswitch.org/11525 I've got my pjsip profile configured for G722 only: Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From anthony.minessale at gmail.com Wed Dec 16 11:26:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 13:26:58 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Message-ID: <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> can you do another trace to show the inbound invite too? On Wed, Dec 16, 2009 at 1:13 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > Pastebin here: > > http://pastebin.freeswitch.org/11525 > > I've got my pjsip profile configured for G722 only: > > > > Yet whenever I send calls using that profile it (mysteriously) > indicates support for PCMU in the INVITE. The pastebin includes both > the INVITE and "sofia status profile pjsip" to show that only G722 has > been enabled. Where is PCMU coming from? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/cebcffec/attachment-0002.html From ranjtech at gmail.com Wed Dec 16 11:30:57 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 14:30:57 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> Message-ID: <01f601ca7e86$4c59c730$e50d5590$@com> MC, haha I'm not sure. I think this had happened to me before as well and nuking the fs dir and then an svn up had fixed it. I think I'll just do an svn co and get on with it. Sorry had been following FS when it first started and then for 2 yrs have been busy with a bunch of random stuff but now want to get back into it. BTW, at that time there was no routing/lcr module available for FS and someone, can't remember who had written one to plug into FS but it wasn't open sourced. Wondering if there is one now to use FS more as a class 5 switch than a PBX for more like a carrier peering and minutes wholesale kind of a business?? Sorry if this is a dumb question :( From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, December 16, 2009 1:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] build errors :( On Wed, Dec 16, 2009 at 10:27 AM, RR wrote: Lost that screen that showed me what rev was downloaded but whatever you get after doing a "svn up" and "make current". I had done a rm -rf in the /usr/src/freeswitch directory and then did svn up. Should I have done svn co instead? Why did you nuke the fs src dir? Just curious. In your case since you deleted everything you're probably better off just starting from scratch with a svn co. Once you're installed then all you need to do is "make current" b/c make current includes an "svn up" among other things. -MC __________ Information from ESET NOD32 Antivirus, version of virus signature database 4694 (20091216) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/d597050a/attachment-0002.html From msc at freeswitch.org Wed Dec 16 11:31:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 11:31:30 -0800 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> Message-ID: <87f2f3b90912161131w69cc5daga3ddab8a33b0654a@mail.gmail.com> Try setting "absolute_codec_string" in the dialplan prior to the bridge: Let us know if that does the trick. -MC On Wed, Dec 16, 2009 at 11:13 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > Pastebin here: > > http://pastebin.freeswitch.org/11525 > > I've got my pjsip profile configured for G722 only: > > > > Yet whenever I send calls using that profile it (mysteriously) > indicates support for PCMU in the INVITE. The pastebin includes both > the INVITE and "sofia status profile pjsip" to show that only G722 has > been enabled. Where is PCMU coming from? > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/0b7b2910/attachment-0002.html From mgg at giagnocavo.net Wed Dec 16 11:32:40 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 16 Dec 2009 14:32:40 -0500 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <26807322.post@talk.nabble.com> References: <26807322.post@talk.nabble.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> We switched to Windows for production after 1.0.4. We've run into no stability issues with it. The highest we go is only 100 sessions/sec. We're also use media bypass and mod_managed. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 Sent: Wednesday, December 16, 2009 12:39 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] [Windows] Stable enough for production use? Hello Since Freeswitch is also available for Windows (and Mac, but I don't anything about Macintosh), I'd like some feedback from users who routinely run Freeswitch on that OS. Is it stable enough to be used in production to handle a single analog line (ie. SOHO use), or should I warn customers that they really should buy a dedicated Linux box to run FS? Thank you. -- View this message in context: http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Dec 16 11:36:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 11:36:57 -0800 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01f601ca7e86$4c59c730$e50d5590$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: <87f2f3b90912161136i2d5bfafeja6b43be1cef8aeab@mail.gmail.com> On Wed, Dec 16, 2009 at 11:30 AM, RR wrote: > MC, haha I?m not sure. I think this had happened to me before as well and > nuking the fs dir and then an svn up had fixed it. > > I think I?ll just do an svn co and get on with it. Sorry had been following > FS when it first started and then for 2 yrs have been busy with a bunch of > random stuff but now want to get back into it. > > > > BTW, at that time there was no routing/lcr module available for FS and > someone, can?t remember who had written one to plug into FS but it wasn?t > open sourced. Wondering if there is one now to use FS more as a class 5 > switch than a PBX for more like a carrier peering and minutes wholesale kind > of a business?? Sorry if this is a dumb question :( > You've got mod_easyroute to handle incoming call routing and mod_lcr for outbound routing. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/daa04363/attachment-0002.html From kristian.kielhofner at gmail.com Wed Dec 16 11:41:00 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:41:00 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> Message-ID: <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> Sure... The call comes up as PCMU: INVITE sip:5888 at 10.70.0.99 SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:9413122830 at smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: , To: "5888" Via: SIP/2.0/UDP 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 Content-Length: 206 Content-Type: application/sdp Contact: Max-Forwards: 70 User-Agent: Avaya CM/R015x.02.0.947.3 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: timer,replaces,join,histinfo,100rel Alert-Info: ;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:9413122830 at smh.sip.local P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: ;index=1,"5888" ;index=1.1 v=0 o=- 1 1 IN IP4 10.70.0.69 s=- c=IN IP4 10.70.0.22 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 We don't support G729 so this call comes up as PCMU when we answer and then that codec is first in the codec list... On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale wrote: > can you do another trace to show the inbound invite too? > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From rupa at rupa.com Wed Dec 16 11:42:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 16 Dec 2009 13:42:51 -0600 Subject: [Freeswitch-users] build errors :( In-Reply-To: <01f601ca7e86$4c59c730$e50d5590$@com> References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: I think you are thinking of SWK. He has checked in mod_easyroute for easy routing of DIDs. [intra]lanman and I have mod_lcr for (umm) lcr routing. SWK still has a (probably) more capable lcr - though not open source. On Wed, Dec 16, 2009 at 1:30 PM, RR wrote: > BTW, at that time there was no routing/lcr module available for FS and > someone, can?t remember who had written one to plug into FS but it wasn?t > open sourced. Wondering if there is one now to use FS more as a class 5 > switch than a PBX for more like a carrier peering and minutes wholesale kind > of a business?? Sorry if this is a dumb question :( -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/c60f2bfc/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 16 11:48:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 13:48:20 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> Message-ID: <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding. if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature. On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Sure... The call comes up as PCMU: > > INVITE sip:5888 at 10.70.0.99 SIP/2.0 > Call-ID: 80ea31a017f6de1d53e4a9c52f00 > CSeq: 1 INVITE > From: sip:9413122830 at smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 > Record-Route: , > To: "5888" > > Via: SIP/2.0/UDP > 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP > > 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 > Content-Length: 206 > Content-Type: application/sdp > Contact: > ;transport=tcp> > Max-Forwards: 70 > User-Agent: Avaya CM/R015x.02.0.947.3 > Allow: > INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH > Supported: timer,replaces,join,histinfo,100rel > Alert-Info: > >;avaya-cm-alert-type=external > Min-SE: 1200 > Session-Expires: 1200;refresher=uac > P-Asserted-Identity: sip:9413122830 at smh.sip.local > P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" > History-Info: > >;index=1,"5888" > >;index=1.1 > > v=0 > o=- 1 1 IN IP4 10.70.0.69 > s=- > c=IN IP4 10.70.0.22 > b=AS:64 > t=0 0 > m=audio 2176 RTP/AVP 18 0 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > > We don't support G729 so this call comes up as PCMU when we answer > and then that codec is first in the codec list... > > On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale > wrote: > > can you do another trace to show the inbound invite too? > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8ceb337d/attachment-0002.html From doddlephone at gmail.com Wed Dec 16 11:42:01 2009 From: doddlephone at gmail.com (Doddle WebPhone) Date: Wed, 16 Dec 2009 17:42:01 -0200 Subject: [Freeswitch-users] Click-to-call and click-to-dial In-Reply-To: References: Message-ID: For an another click2call "flavor/approach", you might use http://www.doddlephone.com to call FS Sergio On Wed, Dec 16, 2009 at 2:24 PM, John Platts wrote: > > You've made my day. > > ________________________________ > > From: jpitcher at nuvio.com > > To: freeswitch-users at lists.freeswitch.org > > Date: Wed, 16 Dec 2009 08:11:05 -0800 > > Subject: Re: [Freeswitch-users] Click-to-call and click-to-dial > > > > > > > > > > > > > > > > > > John, > > > > > > > > To do a click to call in FS you need to have some app that connects to > the ESL or Event Socket Layer and runs one of the calls diagramed here ... > > > > > > > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > > > > > > > For use with the ESL just prepend api in front of the originate so your > call looks something like: > > > > > > > > $command = 'api originate user at domain &bridge(user at domain)'; > > > > > > > > As for programs able to do that from a Microsoft Product, That I am not > sure of. > > > > > > > > Jonathan Pitcher > > > > > > > > > > > > > > > > On 12/16/09 9:59 AM, "John Platts" wrote: > > > > > > > > > > > > > > > > How can I perform click-to-call or click-to-dial in FreeSWITCH? > > > > > > > > Do you have any recommendations on programs capable of click-to-call or > click-to-dial from Microsoft Outlook or Microsoft Excel? > > > > > > > > _________________________________________________________________ > > > > Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. > > > > http://clk.atdmt.com/GBL/go/171222985/direct/01/ > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________ > Hotmail: Trusted email with powerful SPAM protection. > http://clk.atdmt.com/GBL/go/177141665/direct/01/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/1bfad4f3/attachment-0002.html From kristian.kielhofner at gmail.com Wed Dec 16 11:59:17 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 16 Dec 2009 14:59:17 -0500 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> Message-ID: <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> Anthony, As always, thanks. I thought that might be it but I wanted to make sure. Thanks again! On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale wrote: > yah so the codec chosen by the inbound leg is always offered in the outbound > sdp to try and prevent transcoding. > if you set {absolute_codec_string=G722} in the bridge string you will bypass > this feature. > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From ranjtech at gmail.com Wed Dec 16 12:00:44 2009 From: ranjtech at gmail.com (RR) Date: Wed, 16 Dec 2009 15:00:44 -0500 Subject: [Freeswitch-users] build errors :( In-Reply-To: References: <01c901ca7e7a$47a22900$d6e67b00$@com> <112CAA00-83A8-4A22-AEFE-0DD4ACEE1573@freeswitch.org> <01db01ca7e7d$5f14a240$1d3de6c0$@com> <87f2f3b90912161038q6a7232e1s563d6c319d21f9ab@mail.gmail.com> <01f601ca7e86$4c59c730$e50d5590$@com> Message-ID: <020e01ca7e8a$756bb4e0$60431ea0$@com> I think you are thinking of SWK. He has checked in mod_easyroute for easy routing of DIDs. [intra]lanman and I have mod_lcr for (umm) lcr routing. SWK still has a (probably) more capable lcr - though not open source. On Wed, Dec 16, 2009 at 1:30 PM, RR wrote: BTW, at that time there was no routing/lcr module available for FS and someone, can't remember who had written one to plug into FS but it wasn't open sourced. Wondering if there is one now to use FS more as a class 5 switch than a PBX for more like a carrier peering and minutes wholesale kind of a business?? Sorry if this is a dumb question :( -- -Rupa Thanks Rupa and MC! Will check out these modules. I'm hoping/assuming there's some documentation available for these? If not, then would you be able to share some sample configs that work? Thanks \RR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/5a3215f6/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 16 12:16:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 14:16:42 -0600 Subject: [Freeswitch-users] Where is that codec list coming from? In-Reply-To: <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> References: <2d9149cd0912161113i66727b68j421c6c1a1b3dbaa2@mail.gmail.com> <191c3a030912161126o19dd7fecjb42a149ab5395e14@mail.gmail.com> <2d9149cd0912161141m6a2bfebfkd04071e37e26cf88@mail.gmail.com> <191c3a030912161148od0c4b4p8e32968ed6173880@mail.gmail.com> <2d9149cd0912161159i666bd80w5ad9b063ecb75081@mail.gmail.com> Message-ID: <191c3a030912161216o70053862y9c6d3c33be43efe8@mail.gmail.com> np On Wed, Dec 16, 2009 at 1:59 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Anthony, > > As always, thanks. I thought that might be it but I wanted to make sure. > > Thanks again! > > On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale > wrote: > > yah so the codec chosen by the inbound leg is always offered in the > outbound > > sdp to try and prevent transcoding. > > if you set {absolute_codec_string=G722} in the bridge string you will > bypass > > this feature. > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/34ba02af/attachment-0002.html From costa.zikalala at gmail.com Wed Dec 16 12:17:17 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Wed, 16 Dec 2009 22:17:17 +0200 Subject: [Freeswitch-users] Basic Question on the Internal Profile In-Reply-To: <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> References: <59daa2cd0912160437i4d1270e5uc2f6338c092f6782@mail.gmail.com> <87f2f3b90912160909k3390b1c2nefd09b76ef77d341@mail.gmail.com> Message-ID: <59daa2cd0912161217r352942aak94ee04f770b1ced5@mail.gmail.com> Thanks for the detailed response Michael, it certainly helped a lot. 2009/12/16 Michael Collins > > > On Wed, Dec 16, 2009 at 4:37 AM, Costa Zikalala wrote: > >> Hi All >> >> I understand that to connect to a SIP Provider you have to (amongst other >> things) define a Gateway on the External Profile. >> But some gateways may be defined on the Internal Profile. What kind of >> gateways would these be and what would be their purpose as most gateways are >> External? >> >> Hope my question makes a bit of sense. >> >> > This is a good question. First, remember that the profile names "internal" > and "external" are just labels that try to give you a basic idea of what you > might use them for. The external profile is generally used for outbound > registrations, particularly when the FS server is behind NAT. If your FS > server isn't behind NAT then you may want to use the internal profile for > outbound registrations, although we recommend that your outbound > registrations still be in a profile different than your internal profile > where local phones are registering. In these cases it's fine just to remove > external.xml, make a copy of internal.xml to something like > external-nonat.xml, and then edit it so that it uses ports 5080 and 5081. > For the record here's what I did for a user just last week: > > Old: > > > value="$${internal_ssl_enable}"/> > > value="transport=tls"/> > > value="$${internal_tls_port}"/> > > value="$${internal_ssl_dir}"/> > > > > New: > > > value="$${external_ssl_enable}"/> > > value="$${external_tls_port}"/> > > value="$${external_ssl_dir}"/> > > > Feel free to tinker and remember that if you break something you can just > wipe your conf directory and just run "make samples" again and you'll have a > brand new set of default configs. > > Enjoy! > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/99220676/attachment-0002.html From brian at proximosystems.com Wed Dec 16 13:49:09 2009 From: brian at proximosystems.com (Brian) Date: Wed, 16 Dec 2009 16:49:09 -0500 Subject: [Freeswitch-users] mod_conference scalability Message-ID: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8cd81a6d/attachment-0002.html From jerry.richards at teotech.com Wed Dec 16 14:23:11 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 16 Dec 2009 14:23:11 -0800 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 Message-ID: I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" reply. Below is the console loglevel 7 output. It says the destination is out-of-order. I'm not sure what this means. Any help is appreciated. 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0] 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by acl "domains". Falling back to Digest auth. 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0] 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by acl "domains". Falling back to Digest auth. 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel sofia/internal/5381 at 192.168.72.141:5060 [e58e763f-7688-4600-aa70-481bbc359f58] 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: v=0 o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4 192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000/1 a=ptime:20 a=ptime:20 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> CS_INIT 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/5381 at 192.168.72.141:5060) State INIT 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> CS_ROUTING 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_ROUTING 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing Anonymous->93491028 in context default Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->unloop] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->tod_example] continue=true Dialplan: day of week[4] =~ 2-6 (PASS) Dialplan: hour[14] =~ 9-18 (PASS) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(open=true) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->holiday_example] continue=true Dialplan: month[12] =~ 1 (FAIL) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->Mediant1000] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [Mediant1000] destination_number(93491028) =~ /^8(\d+)$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->SangomaPRI] continue=false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [SangomaPRI] destination_number(93491028) =~ /^9(\d+)$/ break=on-false Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(effective_caller_id_number=425740${caller_id_number}) Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action bridge(openzap/smg_prid/a/3491028 at g1) 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> CS_EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(open=true) 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(effective_caller_id_number=4257405381) 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 sofia/internal/5381 at 192.168.72.141:5060 SET [effective_caller_id_number]=[4257405381] EXECUTE sofia/internal/5381 at 192.168.72.141:5060 bridge(openzap/smg_prid/a/3491028 at g1) 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal sofia/internal/5381 at 192.168.72.141:5060 [KILL] 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_HANGUP 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to INVITE with: 502 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> CS_REPORTING 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_REPORTING 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: DESTINATION_OUT_OF_ORDER 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to sleep 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> CS_DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session 1 (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external entities 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session 1 (sofia/internal/5381 at 192.168.72.141:5060) Ended 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close Channel sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep Best Regards, Jerry From brian at freeswitch.org Wed Dec 16 14:27:52 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Dec 2009 16:27:52 -0600 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: References: Message-ID: Need siptrace with this type "sofia profile xxxx siptrace on" replace xxxx with your profile. /b On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote: > I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal > phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" > reply. Below is the console loglevel 7 output. It says the destination is > out-of-order. I'm not sure what this means. Any help is appreciated. > > 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0] > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by > acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0] > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by > acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/5381 at 192.168.72.141:5060 > [e58e763f-7688-4600-aa70-481bbc359f58] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel > sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: > v=0 > o=TC 1100638826 1100638826 IN IP4 192.168.72.32 > s=session > c=IN IP4 192.168.72.32 > t=0 0 > m=audio 1760 RTP/AVP 0 18 4 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000/1 > a=ptime:20 > a=ptime:20 > > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> CS_ROUTING > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING > 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing > Anonymous->93491028 in context default > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing [default->unloop] > continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (PASS) > [tod_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action set(open=true) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match (FAIL) > [holiday_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->Mediant1000] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) [Mediant1000] > destination_number(93491028) =~ /^8(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->SangomaPRI] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) [SangomaPRI] > destination_number(93491028) =~ /^9(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(effective_caller_id_number=425740${caller_id_number}) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 set(open=true) > 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > set(effective_caller_id_number=4257405381) > 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET > [effective_caller_id_number]=[4257405381] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string > 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot create > outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 Originate > Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. > Cause: DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup > sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [KILL] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to INVITE > with: 502 > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to sleep > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session 1 > (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external > entities > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session 1 > (sofia/internal/5381 at 192.168.72.141:5060) Ended > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close Channel > sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From moises.silva at gmail.com Wed Dec 16 15:26:49 2009 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 16 Dec 2009 18:26:49 -0500 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> Message-ID: I've been using FreeSWITCH on Windows lately and seems to work pretty well. Sangoma has been testing more and more lately the Windows drivers with FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 though, always using trunk, if you are going to be using PSTN lines (and therefore requiring openzap) I think it would be a good idea for you to use trunk and latest wanpipe drivers. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com On Wed, Dec 16, 2009 at 2:32 PM, Michael Giagnocavo wrote: > We switched to Windows for production after 1.0.4. We've run into no > stability issues with it. The highest we go is only 100 sessions/sec. We're > also use media bypass and mod_managed. > > -Michael > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 > Sent: Wednesday, December 16, 2009 12:39 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] [Windows] Stable enough for production use? > > > Hello > > Since Freeswitch is also available for Windows (and Mac, but I don't > anything about Macintosh), I'd like some feedback from users who routinely > run Freeswitch on that OS. > > Is it stable enough to be used in production to handle a single analog line > (ie. SOHO use), or should I warn customers that they really should buy a > dedicated Linux box to run FS? > > Thank you. > -- > View this message in context: > http://old.nabble.com/-Windows--Stable-enough-for-production-use--tp26807322p26807322.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/fe91295b/attachment-0002.html From djbinter at yahoo.com Wed Dec 16 15:29:10 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 16 Dec 2009 15:29:10 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite Message-ID: <607753.94827.qm@web37503.mail.mud.yahoo.com> We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/e9398029/attachment-0002.html From anthony.minessale at gmail.com Wed Dec 16 15:42:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Dec 2009 17:42:48 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <607753.94827.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> Message-ID: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and here is the > issue: > > > > Call is initially setup fine and they send a first re-invite with media > 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first > re-invite fine > > > > They then send a second re-invite with their media IP to cut through media > and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it > is strictly for the purpose of ?keep alive? per RFC 4028. This third > re-invite has the exact same media IP and UDP pot information as the second > re-invite does. The problem is FS does not respond to this third re-invite > AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a response from FS. > > > One more thing, we did not see the third re-invite in sofia siptrace, but > we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > > Thank you very much. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/067dd91c/attachment-0002.html From msc at freeswitch.org Wed Dec 16 16:15:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 16:15:38 -0800 Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> And we shouldn't be using 1.0.4 anyway, should we? ;) -MC On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva wrote: > I've been using FreeSWITCH on Windows lately and seems to work pretty well. > Sangoma has been testing more and more lately the Windows drivers with > FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 > though, always using trunk, if you are going to be using PSTN lines (and > therefore requiring openzap) I think it would be a good idea for you to use > trunk and latest wanpipe drivers. > > -- > Moises Silva > Senior Software Engineer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/8ee37d7b/attachment-0002.html From msc at freeswitch.org Wed Dec 16 16:20:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Dec 2009 16:20:12 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00aa01ca7e99$9901f9a0$cb05ece0$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: <87f2f3b90912161620p627d676i85a53c3771d87a91@mail.gmail.com> how are the listeners connecting? On Wed, Dec 16, 2009 at 1:49 PM, Brian wrote: > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/4afd8965/attachment-0002.html From steveu at coppice.org Wed Dec 16 16:48:21 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Dec 2009 08:48:21 +0800 Subject: [Freeswitch-users] What are the solutions for G729 support ? In-Reply-To: <4B29257B.10705@blueface.ie> References: <26777181.post@talk.nabble.com> <191c3a030912140719y3f9949c6g8ba24652ee30e772@mail.gmail.com> <26780206.post@talk.nabble.com> <2B45E9C9-F118-4832-B6D1-0CA91DE7F934@jerris.com> <4B29257B.10705@blueface.ie> Message-ID: <4B297FD5.5060909@coppice.org> On 12/17/2009 02:22 AM, Bruce McAlister wrote: > Hi Michael, > > Michael Jerris wrote: > >> I expect the module will initially be available for linux and we will add other platforms as demand shows a need for it and I can get build servers up that will be used to produce the binaries. Windows will likely be one of the early alternatives >> >> > Is support for Solaris and/or OpenSolaris x86 planned as well? > > Thanks > Bruce > Support for the less popular platforms will probably depend mostly on the availability of machines for building and testing. The code is unlikely to ever be as well optimised for speed on the less popular platforms, but basic support should be largely an issue of access for development and support. Steve From djbinter at yahoo.com Wed Dec 16 17:00:11 2009 From: djbinter at yahoo.com (DJB) Date: Wed, 16 Dec 2009 17:00:11 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> Message-ID: <800257.53977.qm@web37503.mail.mud.yahoo.com> Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: We >have a customer that we are sending calls to off the FS and here is the issue: > >Call >is initially setup fine and they send a first re-invite with media 0.0.0.0 to >place the caller on hold. FS sends a 200 ok to this first re-invite fine > >They >then send a second re-invite with their media IP to cut through media and the >FS sends a 200 OK to this fine. At this point the call is fine > >30 >minutes later they send a third re-invite because according to them it is >strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has >the exact same media IP and UDP pot information as the second re-invite does. >The problem is FS does not respond to this third re-invite AT ALL. It doesn?t >send a 100 trying a 200 OK nothing so this causes the call to be dropped as the >other end does not recieve a response from FS. > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > >Thank you very much. > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/e49d756e/attachment-0002.html From senakahks at gmail.com Wed Dec 16 18:44:31 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Wed, 16 Dec 2009 18:44:31 -0800 (PST) Subject: [Freeswitch-users] Billing solutions information In-Reply-To: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> References: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> Message-ID: <1261017871151-4179366.post@n2.nabble.com> Hi, Seems nobody is interested to talk about this topic. I found nibblebill is great. But doesn't hangup the call when balance goes to zero. The other problem It allows user to call without checking the balance of the cash database. Is this natural?. If this works fine we can easily integrate with a payment gateway like 2checkout. Thank you Lon Baker wrote: > > Hey everyone, > > I am researching billing solutions for Freeswitch and want to consolidate > the information with what others have found, then add it to the Wiki. > > There are seems to be a number of billing solutions by commercial > providers, > claiming they can integrate with Freeswitch, but nothing concrete > explaining > how far they go. > > Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, > etc? > > Mod_nibblebill handles the basics of updating a database table. > > The A2Billing teams says they are planning on adding support for > Freeswitch > in a few months. > > ASTPP.org says they support Freeswitch, but the site hasn't been updated > since 2008. > > If you know about any solutions, links to solutions or any information can > you send it to me? I will organize it and add it to the wiki. > > Thanks! > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Billing-solutions-information-tp4166151p4179366.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mgg at giagnocavo.net Wed Dec 16 20:25:19 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 16 Dec 2009 23:25:19 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <800257.53977.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <800257.53977.qm@web37503.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67032C42866F@mse17be1.mse17.exchange.ms> FWIW, we?ve seen the same thing intermittently, haven?t had time/been able to get a solid repro to capture debug information. Call ID and tags are all matching. After the re-invite fails and the remote end sends a BYE, FS does indeed respond to the re-invite. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of DJB Sent: Wednesday, December 16, 2009 6:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP Re-invite Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091216/ce3e621f/attachment-0002.html From yehavi.bourvine at gmail.com Wed Dec 16 21:16:38 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 07:16:38 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? Message-ID: Hello, I am trying to debug a TLS handshake error between FreeSwitch and some ATA. When setting the loglevel to 9 I get only a message that TLS handshake failed. Is there some other debug command to show what happens during the TLS handshake process? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/c53fbb66/attachment-0002.html From yehavi.bourvine at gmail.com Wed Dec 16 21:39:49 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 07:39:49 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <8b1c9cda0912031417k6cc09ae7w6ca6aef0bf1f4a2c@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: An interim update: - *Audiocodes*: No success yet. I am working with the manufacturer to debug it. - *VegaStream:* Got the necessary license, configured TLS but it doesn't work. I am working with the local representatives on it. Regards, __Yehavi: 2009/12/10 Brian West > I have confirmed it works with Polycom, Snom and a few others .... > polycom is the hardest to set due to having to put the ca cert into > the phone... but other than that its good. > > /b > > On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > > > An intermediate report: > > > > Audiocodes: TLS works only on outgoing requests, incoming ones are > > ignored. I am waiting for Audiocodes' help in order to debug it. > > SRTP: worked when no TLS is active. When TLS is active the call is > > disconnected when the remote party answers. Still debugging it. > > > > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > > how to enable TLS from the WEB interface. > > > > Regards, __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/708106d7/attachment-0002.html From juanbackson at gmail.com Wed Dec 16 21:53:36 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 17 Dec 2009 13:53:36 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> Message-ID: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Hi I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. Is there any other ways to check for zombie channels? jb On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: > Why not just set rtp-timeout-sec on the sofia profile and it'll do > that for you. > > Unless something else is going on. > > /b > > On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > > > Hi, > > > > I am having problem with around 1 % of the channels always get > > zombilized. > > > > What I want to do is to have a background thread that regularly > > check all the channels that have been in existance for like > 1 hr, > > and then check to see if there is any RTP coming in and going out. > > If there is no RTP, then I just hangup that channel. Does anyone > > know if there is anyway to do that in a freeswitch module? Which > > API can I use to accomplish this purpose? Alternatively, is there > > anyway to configure freeswitch so that it will hangup the calls > > where there is no media in and out for so many seconds? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7cdefe03/attachment-0002.html From mcampbellsmith at gmail.com Wed Dec 16 22:39:56 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 17 Dec 2009 17:39:56 +1100 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <33c87fa30912031434v140060d4j3d0ebf816bfee845@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> Message-ID: <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> Thanks Yehavi... I posted a question on the Cisco Forum and am waiting a response from 'engineering' to tell us if they plan to implement standard SRTP support in the Linksys ATA's. TLS is working fine. On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine wrote: > An interim?update: > > > Audiocodes: No success yet.?I am working with the manufacturer to debug it. > VegaStream: Got the necessary license, configured TLS but it doesn't work. I > am working with the local representatives on it. > > ????????????????????????????? Regards, __Yehavi: > > 2009/12/10 Brian West >> >> I have confirmed it works with Polycom, Snom and a few others .... >> polycom is the hardest to set due to having to put the ca cert into >> the phone... but other than that its good. >> >> /b >> >> On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: >> >> > An intermediate report: >> > >> > Audiocodes: TLS works only on outgoing requests, incoming ones are >> > ignored. I am waiting for Audiocodes' help in order to debug it. >> > SRTP: worked when no TLS is active. When TLS is active the call is >> > disconnected when the remote party answers. Still debugging it. >> > >> > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions >> > how to enable TLS from the WEB interface. >> > >> > ? ? ? ? ? ? ? ? ? ? ? ? ?Regards, __Yehavi: >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From yehavi.bourvine at gmail.com Wed Dec 16 23:29:43 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 09:29:43 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). Any idea whether it is possible to program Freeswitch to support this draft? Thanks, __Yehavi: 2009/11/29 Ujjval Karihaloo > Polycom Firmware matrix (Look at the polycom website) does not allow > firmware higher than 2.3.2 (I think) to be loaded on the old 501 phones?So > first confirm you are on a supported firmware release? > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Yehavi > Bourvine > *Sent:* Sunday, November 29, 2009 8:48 AM > *To:* freeswitch-users > *Subject:* [Freeswitch-users] Polycom 501 conferencing with FreeSwitch > > > > Hello, > > > > I am trying to set a Polycom 501 phone to do conferencing via the > conference room on Freeswitch rather than on the phone (as on the phone it > is limited to 3 participants only). Anyone had success with it? > > > > I have on the Freeswitch an extension named Conf.* which activates the > conference application (it works with other brands). On the Polycom I tried > to define > > voIpProt.SIP.*conference*.address=sip:Conf0000 at freeswitch-server. The > phone continues to create the conference locally and add the above Conf0000 > to it, without REFERing the parties to it. The first phone which called is > left on hold... > > > > Anyone managed to make this feature work? We use firmware 3.1.3. > > > > Thanks! __Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/a57eba73/attachment-0002.html From talk2ram at gmail.com Thu Dec 17 02:29:10 2009 From: talk2ram at gmail.com (ram) Date: Thu, 17 Dec 2009 02:29:10 -0800 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: Hi Look at Contrib of source http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ some pre-paid examples Ram On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi wrote: > Dear Sir, > > I have successfully installed freeSWITCH and it works fine in passthrough > mode. I installed nibblebill and it deduct money from the accounts database > and it works fine. but I have two problems. > > 1. Calls can be initiated even though there is a minus value in accounts > database > > 2. Calls doesn't hangup when it goes to minus values. > > Any answers are greatly appreciated. > > This is my dialplan: > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/9e1ae678/attachment-0002.html From yivzhenko at mksat.net Thu Dec 17 03:05:56 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Thu, 17 Dec 2009 13:05:56 +0200 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 Message-ID: <200912171305.57498.yivzhenko@mksat.net> Hallo All! I need information about setup BLF on GXP2010/2020 phones with Freeswitch. I search in Freeswitch Wiki and maillist archives but find no usable information. From oscav at hotmail.fr Thu Dec 17 03:21:21 2009 From: oscav at hotmail.fr (Oscav) Date: Thu, 17 Dec 2009 03:21:21 -0800 (PST) Subject: [Freeswitch-users] How to set the Session Name on a SDP? In-Reply-To: <26815554.post@talk.nabble.com> References: <26815554.post@talk.nabble.com> Message-ID: <26826579.post@talk.nabble.com> I just found that this is related to the username of the profile. It needs to be set as parameter. Oscav wrote: > > Hi, > > Is it possible to set (rewrite) the Session Name in the SDP of a 183 > progress sent to inbound ? > > Many thanks > -- View this message in context: http://old.nabble.com/How-to-set-the-Session-Name-on-a-SDP--tp26815554p26826579.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 17 03:41:54 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 06:41:54 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> Message-ID: <4B2A1902.2050008@aastral.net> Okay, I added: to my sofia profile and restarted sofia, and still no joy. I'm on FreeSWITCH Version 1.0.trunk (15764) I've got in the directory, but I'm still being rejected by the acl: 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.103.12/32 Here's what I believe is the appropriate snippet of the debug output: http://pastebin.freeswitch.org/11531 Thoughts? Thanks, Bill Brian West wrote: > use "apply-proxy-acl" on the sofia profile. > > /b > > On Dec 15, 2009, at 10:58 PM, Bill W wrote: > >> However, having the proxy in the path effectively negates using IP >> based >> ACLS. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From senakahks at gmail.com Thu Dec 17 03:53:03 2009 From: senakahks at gmail.com (Senaka Amarakeerthi) Date: Thu, 17 Dec 2009 20:53:03 +0900 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: Dear Ram, Thank you for the reply. To work with your code I hope that Mod cdr should be there. But wiki says that its not functional. What should I do. Thanks Senaka On Thu, Dec 17, 2009 at 7:29 PM, ram wrote: > Hi > > Look at Contrib of source > > http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/diegoviola/ > > some pre-paid examples > > Ram > > On Wed, Dec 16, 2009 at 12:27 AM, Senaka Amarakeerthi > wrote: >> >> Dear Sir, >> >> I have successfully installed freeSWITCH and it works fine in passthrough >> mode. I installed nibblebill and it deduct money from the accounts >> database >> and it works fine. but I have two problems. >> >> 1. Calls can be initiated even though there is a minus value in accounts >> database >> >> 2. Calls doesn't hangup when it goes to minus values. >> >> Any answers are greatly appreciated. >> >> This is my dialplan: >> >> >> >> >> ? >> ? ? >> ? ? >> ? >> >> ? >> >> >> >> >> >> > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> >> >> >> >> >> >> This is the configuration file; >> >> >> ? >> ? ? >> >> ? ? >> >> >> >> >> ? ? >> >> >> ? ? >> >> >> ? ? >> >> >> >> ? ? >> >> >> ? ? >> >> >> >> ? ? >> >> >> >> ? ? >> >> >> >> ? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From saeedahmad1981 at gmail.com Thu Dec 17 03:54:45 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 17 Dec 2009 12:54:45 +0100 Subject: [Freeswitch-users] Sofia performance In-Reply-To: <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> References: <23f91030912130747y74417055u1494287e0c569e24@mail.gmail.com> <20091213162932.GW31924@base.carmickle.com> <191c3a030912131036m2f937778ja65d87036b143f7@mail.gmail.com> <8D3D2008-5301-4BDB-9D65-1F2134DC68F9@gmail.com> <191c3a030912131437p17ee7c87gf96ee04d82205deb@mail.gmail.com> Message-ID: with the scenario below can we get the better performance: We create one profile for incoming call listening on 5060 as profile1 we create two profile for outgoing calls as profile2 on 5050 and profile3 on 5051 now we are receiving all calls on profile1:5060, but while bridging them to vendors we divide them, half to profile2:5050 and half to profile3:5051, something like: Will it make any difference? Thanks On Sun, Dec 13, 2009 at 11:37 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Sep processes does better than sep profiles. We need to push the sofia > devs to work on a better concurrancy scheme but they are too busy with other > nokia duties these days so were stuck with what we got for now. About > 400cps on a good day > > On Dec 13, 2009 4:05 PM, "Jay Binks" wrote: > > I'm interested in what the upper limit would be, when expecting a > performance improvement with sofia profiles. > > For example let's say I were to direct connect to customers ( layer 2 ) > with a .1q trunk coming in to fs and a Sofia profile for each customer. Am > I going to hit a bottleneck at 20,50,100,500 ??? > > Guess it's hardware limited , but any thoughts ? > > J > > On 14/12/2009, at 4:36, Anthony Minessale > wrote: > Here is my standa... > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/37563e6f/attachment-0002.html From codecomplete at free.fr Thu Dec 17 04:54:49 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 04:54:49 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <72220.45962.qm@web111310.mail.gq1.yahoo.com> References: <26808383.post@talk.nabble.com> <72220.45962.qm@web111310.mail.gq1.yahoo.com> Message-ID: <26827581.post@talk.nabble.com> I don't have access to a remote computer from which I could log on and run nmap. I'll see if I can get a shell access somewhere. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From foxb at abv.bg Thu Dec 17 05:14:41 2009 From: foxb at abv.bg (Hristo Benev) Date: Thu, 17 Dec 2009 15:14:41 +0200 (EET) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? Message-ID: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Just for your information there is a version of nmap for windows. So you can do the test from your desktop. >-------- ?????????? ????? -------- >??: Fred-145 >???????: Re: [Freeswitch-users] Scanning my firewall for open UDP ports? >??: freeswitch-users at lists.freeswitch.org >????????? ??: ?????????, 2009, ???????? 17 14:54:49 EET > >I don't have access to a remote computer from which I could log on and run >nmap. > >I'll see if I can get a shell access somewhere. Thank you. >-- >View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26827581.html >Sent from the Freeswitch-users mailing list archive at Nabble.com. > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > From neilp at cs.stanford.edu Thu Dec 17 05:34:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 17 Dec 2009 19:04:58 +0530 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: Hi Mike, This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. In case this wasn't apparent I am trying to install FS from trunk. Thanks, Neil On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote: > strange, can someone file a bug on this on jira.freeswitch.org and contact > me off list with ssh info so I can troubleshoot this on your box. > > Thanks > Mike > > On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > > I'm also experiencing this problem, and I have verified I have libogg, > libvorbis, and their dev packages installed. > > I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not > listed in the dependency lib list. Is this related? > > -Neil > > On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: > >> looks like ogg devel packages are installed but ogg lib is not? >> >> >> On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: >> >> > FreeSWITCH seems to be unable to read MP3 files, citing that it's an >> > unknown format. Looking through the log, I found this during startup: >> > >> > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error >> > Loading module /usr/local/freeswitch/mod/mod_shout.so >> > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: >> > ogg_sync_wrote** >> > >> > There don't seem to be any compile-time errors, yet I can't seem to >> > eliminate this issue. Any help would be appreciated. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/1cf3832c/attachment-0002.html From brian at freeswitch.org Thu Dec 17 06:50:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:50:35 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Message-ID: <7C2A7FEA-BB01-4176-B64D-776C40565F01@freeswitch.org> We need more info... svn rev, gcore, back trace and what not... please see the reporting bugs link on the wiki. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Dec 16, 2009, at 11:53 PM, Juan Backson wrote: > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls with duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb From brian at freeswitch.org Thu Dec 17 06:51:05 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:51:05 -0600 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2A1902.2050008@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> Message-ID: <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> it needs to be an ACL from acl.conf or a ip/cidr /b On Dec 17, 2009, at 5:41 AM, Bill W wrote: > Okay, I added: to my sofia > profile and restarted sofia, and still no joy. > > I'm on FreeSWITCH Version 1.0.trunk (15764) > I've got in > the directory, but I'm still being rejected by the acl: > > 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 > Rejected by user acl 190.218.103.12/32 > > Here's what I believe is the appropriate snippet of the debug output: > http://pastebin.freeswitch.org/11531 > > Thoughts? > Thanks, > Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/aba9c27b/attachment-0002.html From brian at freeswitch.org Thu Dec 17 06:51:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 08:51:27 -0600 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: Works on my CentOS 5.4 box just fine... /b On Dec 17, 2009, at 7:34 AM, Neil Patel wrote: > Hi Mike, > > This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. > > In case this wasn't apparent I am trying to install FS from trunk. > > Thanks, > Neil From brian at freeswitch.org Thu Dec 17 07:00:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 09:00:54 -0600 Subject: [Freeswitch-users] How to set the Session Name on a SDP? In-Reply-To: <26826579.post@talk.nabble.com> References: <26815554.post@talk.nabble.com> <26826579.post@talk.nabble.com> Message-ID: <3488E7DE-395B-41A4-A65D-73C0ACC33358@freeswitch.org> Why are you needing to change it? /b On Dec 17, 2009, at 5:21 AM, Oscav wrote: > > I just found that this is related to the username of the profile. It needs to > be set as parameter. From mike at jerris.com Thu Dec 17 07:18:25 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:18:25 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00aa01ca7e99$9901f9a0$cb05ece0$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn?t scale nearly as well as I?d hoped (based on what I?ve read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I?ve set file logging to ?notice? level. My conference profile is configured to suppress several events, hoping that it would improve performance. > > Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: > > Scenario 1: > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/d8361473/attachment-0002.html From mike at jerris.com Thu Dec 17 07:36:44 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:36:44 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <607753.94827.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> Message-ID: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and here is the issue: > > > > Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. > > > > One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ce45758c/attachment-0002.html From mike at jerris.com Thu Dec 17 07:39:03 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:39:03 -0500 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: Its software, anything is possible with enough time and effort. Mike On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). > > Any idea whether it is possible to program Freeswitch to support this draft? > > Thanks, __Yehavi: > From djbinter at yahoo.com Thu Dec 17 07:48:54 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 07:48:54 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> Message-ID: <206417.45016.qm@web37507.mail.mud.yahoo.com> Anthony, I have pasted the invite sip trace here:? http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: >? >Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >? >They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >? >30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > >Thank you very much. > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/636afbe0/attachment-0002.html From mayamatakeshi at gmail.com Thu Dec 17 07:53:54 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 18 Dec 2009 00:53:54 +0900 Subject: [Freeswitch-users] Small delay in registration validity Message-ID: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> It seems to me, in previous revisions of FS, we could successfully call a registered user as soon as his terminal gets 200 OK for REGISTER. But after testing recent revisions, it seems we must wait a little (I wait 1 second) otherwise a call to bridge would end with this: 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] Similar thing is happening when the terminal unregisters: after unregistration an immediate call to bridge sofia/profile/user%domain will succeed. Has anything changed recently in the way registration works that could explain this? br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/fc45cb14/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 07:54:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 09:54:21 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> Message-ID: <191c3a030912170754l4589fb94v190820b51c39bada@mail.gmail.com> Is the packet capture running on the FS box itself? On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris wrote: > if you don't see it in sofia siptrace but do see it in tcpdump capture then > something very ugly is going on. Either sofia has hung up completely and is > not listening on that port anymore (can other calls go through?) or the > packet you see in tcpdump is not really going to the right port. Can you > confirm which one? > > Mike > > On Dec 16, 2009, at 6:29 PM, DJB wrote: > > We have a customer that we are sending calls to off the FS and here is the > issue: > > > > Call is initially setup fine and they send a first re-invite with media > 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first > re-invite fine > > > > They then send a second re-invite with their media IP to cut through media > and the FS sends a 200 OK to this fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because according to them it > is strictly for the purpose of ?keep alive? per RFC 4028. This third > re-invite has the exact same media IP and UDP pot information as the second > re-invite does. The problem is FS does not respond to this third re-invite > AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a response from FS. > > > One more thing, we did not see the third re-invite in sofia siptrace, but > we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b91a98b0/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 07:57:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 09:57:42 -0600 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > ------------------------------ > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wed, December 16, 2009 3:42:48 PM > *Subject:* Re: [Freeswitch-users] SIP Re-invite > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >> We have a customer that we are sending calls to off the FS and here is >> the issue: >> >> >> >> Call is initially setup fine and they send a first re-invite with media >> 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first >> re-invite fine >> >> >> >> They then send a second re-invite with their media IP to cut through media >> and the FS sends a 200 OK to this fine. At this point the call is fine >> >> >> >> 30 minutes later they send a third re-invite because according to them it >> is strictly for the purpose of ?keep alive? per RFC 4028. This third >> re-invite has the exact same media IP and UDP pot information as the second >> re-invite does. The problem is FS does not respond to this third re-invite >> AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the >> call to be dropped as the other end does not recieve a response from FS. >> >> >> One more thing, we did not see the third re-invite in sofia siptrace, but >> we do see it in ethereal, which is kind of odds. >> >> >> We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >> Thank you very much. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/6a1d4b55/attachment-0002.html From mike at jerris.com Thu Dec 17 07:58:42 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 10:58:42 -0500 Subject: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote In-Reply-To: References: <9E40BA85-3F55-4AD0-9322-78D02244CF1F@seanf.me> <5D7165BF-9089-4A72-BCA0-7DA50400B118@jerris.com> Message-ID: <5CB306F6-3868-41CA-B633-F7809D1540B8@jerris.com> if you contact me offlist, or better, join #freeswitch on irc.freenode.net and ping me (MikeJ) Mike On Dec 17, 2009, at 8:34 AM, Neil Patel wrote: > Hi Mike, > > This has shown up on my laptop running ubuntu karmic. If we plan ahead, I can setup ssh access for you to check things out. > > In case this wasn't apparent I am trying to install FS from trunk. > > Thanks, > Neil > > On Wed, Dec 16, 2009 at 9:14 PM, Michael Jerris wrote: > strange, can someone file a bug on this on jira.freeswitch.org and contact me off list with ssh info so I can troubleshoot this on your box. > > Thanks > Mike > > On Dec 16, 2009, at 9:56 AM, Neil Patel wrote: > >> I'm also experiencing this problem, and I have verified I have libogg, libvorbis, and their dev packages installed. >> >> I looked at mod/mod_shout.la and noticed that '/usr/lib/libogg' is not listed in the dependency lib list. Is this related? >> >> -Neil >> >> On Sat, Nov 7, 2009 at 11:22 PM, Michael Jerris wrote: >> looks like ogg devel packages are installed but ogg lib is not? >> >> >> On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: >> >> > FreeSWITCH seems to be unable to read MP3 files, citing that it's an >> > unknown format. Looking through the log, I found this during startup: >> > >> > 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error >> > Loading module /usr/local/freeswitch/mod/mod_shout.so >> > **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: >> > ogg_sync_wrote** >> > >> > There don't seem to be any compile-time errors, yet I can't seem to >> > eliminate this issue. Any help would be appreciated. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/5e13de31/attachment-0002.html From mrene_lists at avgs.ca Thu Dec 17 08:00:12 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 17 Dec 2009 11:00:12 -0500 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> Message-ID: <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> Are you doing proxy or bypass meda? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Dec-09, at 12:53 AM, Juan Backson wrote: > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls > with duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb > > On Wed, Dec 16, 2009 at 10:52 PM, Brian West > wrote: > Why not just set rtp-timeout-sec on the sofia profile and it'll do > that for you. > > Unless something else is going on. > > /b > > On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: > > > Hi, > > > > I am having problem with around 1 % of the channels always get > > zombilized. > > > > What I want to do is to have a background thread that regularly > > check all the channels that have been in existance for like > 1 hr, > > and then check to see if there is any RTP coming in and going out. > > If there is no RTP, then I just hangup that channel. Does anyone > > know if there is anyway to do that in a freeswitch module? Which > > API can I use to accomplish this purpose? Alternatively, is there > > anyway to configure freeswitch so that it will hangup the calls > > where there is no media in and out for so many seconds? > > > > Thanks, > > jb > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7ad25dcb/attachment-0002.html From mike at jerris.com Thu Dec 17 08:03:46 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 11:03:46 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/170d35d9/attachment-0002.html From kristian.kielhofner at gmail.com Thu Dec 17 08:04:02 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 11:04:02 -0500 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: Message-ID: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> You could try ssldump: http://www.rtfm.com/ssldump/ On Thu, Dec 17, 2009 at 12:16 AM, Yehavi Bourvine wrote: > Hello, > > ? I am trying to debug a TLS handshake error between FreeSwitch and some > ATA.?When setting the loglevel to 9 I get only a message that TLS handshake > failed. Is there some other debug command to show what happens during the > TLS handshake process? > > ??????????????????????????? Thanks! __Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From yehavi.bourvine at gmail.com Thu Dec 17 08:07:41 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 18:07:41 +0200 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... Thanks, __Yehavi: 2009/12/17 Michael Jerris > Its software, anything is possible with enough time and effort. > > Mike > > On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > > > After some discussions with Polycom support it seems that their > conferencing support is based on draft-ietf-sipping-cc-conferencing-03 > (which is not the latest and is not compatible with the latest one). > > > > Any idea whether it is possible to program Freeswitch to support this > draft? > > > > Thanks, __Yehavi: > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7a60083a/attachment-0002.html From brian at freeswitch.org Thu Dec 17 08:08:38 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 10:08:38 -0600 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> Message-ID: <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Also what device are you using? I haven't tested with many so far... Polycom, Snom and a few others do TLS (see interop page on wiki) others do it wrong. /b On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: > You could try ssldump: > > http://www.rtfm.com/ssldump/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/42dccdc8/attachment-0002.html From djbinter at yahoo.com Thu Dec 17 08:11:34 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 08:11:34 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> Message-ID: <556038.39248.qm@web37502.mail.mud.yahoo.com> The trace that I pasted on the pastebin was from?our analyzer,Tektronix spectra2 that was sitting between FS and customer.? I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here:? http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >>? >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >>? >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >>? >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >pstn:?+19193869900? +19193869900 > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/eea48cca/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 08:12:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 10:12:58 -0600 Subject: [Freeswitch-users] Small delay in registration validity In-Reply-To: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> References: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> Message-ID: <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> The sql is sorted into transactions to boost performance so it waits for either 500 statements to execute or 500ms to elapse to accumulate as many sql stmts as possible into the transaction. set sql-in-transactions to false in your profile or make a patch to make the 500ms configurable On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote: > It seems to me, in previous revisions of FS, we could successfully call a > registered user as soon as his terminal gets 200 OK for REGISTER. > But after testing recent revisions, it seems we must wait a little (I wait > 1 second) otherwise a call to bridge would end with this: > > 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create > outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] > > Similar thing is happening when the terminal unregisters: after > unregistration an immediate call to bridge sofia/profile/user%domain will > succeed. > > Has anything changed recently in the way registration works that could > explain this? > > br, > takeshi > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/18f2f840/attachment-0002.html From dave at 3c.co.uk Thu Dec 17 08:13:30 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 16:13:30 +0000 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <206417.45016.qm@web37507.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> Message-ID: <1261066410.6396.73.camel@local.freepabx.com> I'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > We have a customer that we are sending calls to off the FS and > here is the issue: > > > > Call is initially setup fine and they send a first re-invite > with media 0.0.0.0 to place the caller on hold. FS sends a 200 > ok to this first re-invite fine > > > > They then send a second re-invite with their media IP to cut > through media and the FS sends a 200 OK to this fine. At this > point the call is fine > > > > 30 minutes later they send a third re-invite because according > to them it is strictly for the purpose of ?keep alive? per RFC > 4028. This third re-invite has the exact same media IP and UDP > pot information as the second re-invite does. The problem is > FS does not respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the call to > be dropped as the other end does not recieve a response from > FS. > > > One more thing, we did not see the third re-invite in sofia > siptrace, but we do see it in ethereal, which is kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in bypass > media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 17 08:13:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 10:13:41 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> Message-ID: <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> sip session timers is the standardized way to handle this. On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene wrote: > Are you doing proxy or bypass meda? > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Dec-09, at 12:53 AM, Juan Backson wrote: > > Hi > > I have rtp-timeout-sec set to 300 s but I am still getting calls with > duration of 1 day long. > > Is there any other ways to check for zombie channels? > > jb > > On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: > >> Why not just set rtp-timeout-sec on the sofia profile and it'll do >> that for you. >> >> Unless something else is going on. >> >> /b >> >> On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: >> >> > Hi, >> > >> > I am having problem with around 1 % of the channels always get >> > zombilized. >> > >> > What I want to do is to have a background thread that regularly >> > check all the channels that have been in existance for like > 1 hr, >> > and then check to see if there is any RTP coming in and going out. >> > If there is no RTP, then I just hangup that channel. Does anyone >> > know if there is anyway to do that in a freeswitch module? Which >> > API can I use to accomplish this purpose? Alternatively, is there >> > anyway to configure freeswitch so that it will hangup the calls >> > where there is no media in and out for so many seconds? >> > >> > Thanks, >> > jb >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ddb3789f/attachment-0002.html From stevesteffler at shaw.ca Thu Dec 17 08:13:15 2009 From: stevesteffler at shaw.ca (Steve Steffler) Date: Thu, 17 Dec 2009 09:13:15 -0700 Subject: [Freeswitch-users] mod_voicemail question In-Reply-To: <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> References: <451A199B-E2E9-4BCA-87A0-DF853950F9BB@shaw.ca> <87f2f3b90912151109q204385d0i50c87e69964d4d4@mail.gmail.com> Message-ID: <1266CBCF-A246-4776-84CA-686CBCA93EA0@shaw.ca> Hello Micheal On Dec 15, 2009, at 12:09 PM, Michael Collins wrote: > Hi all, > > What is the difference between the mod_voicemail "vm_message_ext" parameter and the "file-extension" parameter? > > vm_message_ext is a channel variable: > http://wiki.freeswitch.org/wiki/Mod_voicemail#vm_message_ext > > file-extension is a parameter of the voicemail module: > http://wiki.freeswitch.org/wiki/Mod_voicemail#file-extension > > The former sets for a specific user, the latter for mod_voicemail in general. Ahh, thanks for clearing this up for me! Now I understand the difference. > > I want all my voicemail in .WAV format except for a couple of extensions which need to be in MP3. > > I'm getting strange results playing with these settings, for example, after logging into the voicemail, it will say "You have 1 new message. First message at ", and then instead of the voicemail message there will be silence and a long pause. Then it will repeat the message count and loop this behavior. During the silence, I seem to be able to press keys to trigger voicemail events, like for example I am allowed to delete the message (although it isn't playing the message to me, and I am instead hearing silence). > > Any ideas? > > Is this perhaps a recording of silence, so that you might actually be listening to a message? > -MC Nope, turns out that according to the FS logs it was trying to play a {uuid}.WAV file that it was expecting to still be there, but which was deleted from a previous checking of voicemail. It's like the database of new messages was out of sync with the message sound files in the mailbox on the server. I deleted the 'ghost' voicemails from my mailbox and now things are back to normal. That could have been a result of my experimentations, I doubt it was a problem with FS. Thanks for your help and keep up the outstanding work! I love FreeSWITCH. :-) Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/50fd7f91/attachment-0002.html From os at tenios.de Thu Dec 17 08:13:18 2009 From: os at tenios.de (=?iso-8859-1?Q?Oliver_Sch=F6nbeck?=) Date: Thu, 17 Dec 2009 17:13:18 +0100 Subject: [Freeswitch-users] Voicemail->Email Message-ID: <010101ca7f33$d9359eb0$8ba0dc10$@de> Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>&1 [[ -t 1 ]] && echo "Writing to logfile '$LOG'." exec > $LOG 2>&1 exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/971ab953/attachment-0002.html From brian at freeswitch.org Thu Dec 17 08:17:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 10:17:15 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <010101ca7f33$d9359eb0$8ba0dc10$@de> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> Message-ID: <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: > Hello, > > we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. > > So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > typeset LOG="/tmp/${0##*/}.out" > mv $LOG ${LOG}.old >/dev/null 2>&1 > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > exec > $LOG 2>&1 > exim4 -t -v >> $LOG > > If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG > > Has anybody seen similar effects before? > > Any advice whats going wrong is heavily appreciated. > > Thanks > Oliver > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/556ebb4b/attachment-0002.html From djbinter at yahoo.com Thu Dec 17 08:19:32 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 08:19:32 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <93BC9BC0-88F2-4163-8115-7D28E393B617@jerris.com> Message-ID: <446963.62507.qm@web37507.mail.mud.yahoo.com> It only happened to the calls from this customer that keeps sending re-invite every 30 minutes, since their switch is expecting a reply back from those re-invite and FS did not respond back to those re-invite. Thank you.? ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:36:44 AM Subject: Re: [Freeswitch-users] SIP Re-invite if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. ?Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. ?Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: >? >Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >? >They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >? >30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. ? > > >One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. > > >We are running?FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/37dd7443/attachment-0002.html From os at tenios.de Thu Dec 17 08:33:58 2009 From: os at tenios.de (=?iso-8859-1?Q?Oliver_Sch=F6nbeck?=) Date: Thu, 17 Dec 2009 17:33:58 +0100 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> <7FDF9B6E-9D45-4B1C-A920-5658171F66E8@freeswitch.org> Message-ID: <014201ca7f36$bbc371b0$334a5510$@de> Currently it is Version 1.0.trunk (15982) Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West Gesendet: Donnerstag, 17. Dezember 2009 17:17 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Voicemail->Email What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>&1 [[ -t 1 ]] && echo "Writing to logfile '$LOG'." exec > $LOG 2>&1 exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b54aad96/attachment-0002.html From mayamatakeshi at gmail.com Thu Dec 17 08:50:48 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 18 Dec 2009 01:50:48 +0900 Subject: [Freeswitch-users] Small delay in registration validity In-Reply-To: <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> References: <15b9404e0912170753v7cafbfdcj748f3811ee9f0ada@mail.gmail.com> <191c3a030912170812w28c37206u937e24b1778ebcf1@mail.gmail.com> Message-ID: <15b9404e0912170850w348fb16at4dfafa7915207506@mail.gmail.com> On Fri, Dec 18, 2009 at 1:12 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The sql is sorted into transactions to boost performance so it waits for > either 500 statements to execute or 500ms to elapse to accumulate as many > sql stmts as possible into the transaction. > > set sql-in-transactions to false in your profile or make a patch to make > the 500ms configurable > Thanks. To change the param sql-in-transactions is enough for me (just during tests). I tested setting it to false and the behavior is as expected. I have updated the wiki: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#sql-in-transactions > On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote: > >> It seems to me, in previous revisions of FS, we could successfully call a >> registered user as soon as his terminal gets 200 OK for REGISTER. >> But after testing recent revisions, it seems we must wait a little (I wait >> 1 second) otherwise a call to bridge would end with this: >> >> 2009-12-17 22:46:14.155163 [ERR] switch_ivr_originate.c:2249 Cannot create >> outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] >> >> Similar thing is happening when the terminal unregisters: after >> unregistration an immediate call to bridge sofia/profile/user%domain will >> succeed. >> >> Has anything changed recently in the way registration works that could >> explain this? >> >> br, >> takeshi >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c289dc0c/attachment-0002.html From codecomplete at free.fr Thu Dec 17 08:56:30 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 08:56:30 -0800 (PST) Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? Message-ID: <26831043.post@talk.nabble.com> Hello I'm no wget expert, and figured I should ask here first: I'd like to download the whole wiki using wget for off-line reading. Using the following didn't work: wget -m -np http://wiki.freeswitch.org/wiki/Main_Page If I move the wiki/ directory to the root directory of my web server, and try to open http://localhost/wiki/Main_Page, FireFox tries to download the page with this dialog box: "You have chosen to open Main_Page which is a: application/octet-stream" I assume wget can do this, but I don't know enough. Has someone succeeded in downloading the whole wiki with wget and could give the right switches to use? Thank you. -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831043.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 17 09:00:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 11:00:52 -0600 Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? In-Reply-To: <26831043.post@talk.nabble.com> References: <26831043.post@talk.nabble.com> Message-ID: I would rather you not do that with wget you beat the hell out of the wiki resources... how often do you do this? I would try doing a printable version. /b On Dec 17, 2009, at 10:56 AM, Fred-145 wrote: > > Hello > > I'm no wget expert, and figured I should ask here first: I'd like to > download the whole wiki using wget for off-line reading. > > Using the following didn't work: > > wget -m -np http://wiki.freeswitch.org/wiki/Main_Page > > If I move the wiki/ directory to the root directory of my web server, and > try to open http://localhost/wiki/Main_Page, FireFox tries to download the > page with this dialog box: > > "You have chosen to open > Main_Page > which is a: application/octet-stream" > > I assume wget can do this, but I don't know enough. Has someone succeeded in > downloading the whole wiki with wget and could give the right switches to > use? > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3604e24b/attachment-0002.html From stevendt at primrosebank.net Thu Dec 17 09:02:10 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 17 Dec 2009 17:02:10 -0000 Subject: [Freeswitch-users] Building on Windows Message-ID: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) 5. How do I specify which options (e.g., mod_flite, to be included iin the build. 6. How do I build the sounds etc. ? Regards Dave 5>os_unix.obj : warning LNK4221: no public symbols found; archive member will be inaccessible 1>filesys.c 6>..\..\pcre\pcre_ucp_searchfuncs.c(158) : warning C4018: '<' : signed/unsigned mismatch 6>..\..\pcre\pcre_ucp_searchfuncs.c(163) : warning C4018: '<=' : signed/unsigned mismatch 16>getenv.c 15>..\..\src\switch_console.c(553) : warning C4244: '=' : conversion from '__w64 int' to 'int', possible loss of data 15>..\..\src\switch_console.c(584) : warning C4267: '=' : conversion from 'size_t' to 'int', possible loss of data 15>switch_core_media_bug.c 15>..\..\src\switch_core_media_bug.c(178) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(221) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 15>..\..\src\switch_core_media_bug.c(222) : warning C4244: '=' : conversion from 'switch_size_t' to 'uint32_t', possible loss of data 18>nta_tag.c 21>c:\freeswitch src\freeswitch\libs\xmlrpc-c\src\xmlrpc_server_abyss.c(894) : warning C4047: 'initializing' : 'TOsSocket' differs in levels of indirection from 'void *' 31>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>cl : Command line warning D9040 : ignoring option '/analyze'; Code Analysis warnings are not available in this edition of the compiler 86>mod_opal.cpp 86>c:\freeswitch src\freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error C1083: Cannot open include file: 'ptlib.h': No such file or directory -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/fcbacae3/attachment-0002.html From codecomplete at free.fr Thu Dec 17 09:14:21 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 17 Dec 2009 09:14:21 -0800 (PST) Subject: [Freeswitch-users] Mirroring wiki with wget for offline browsing? In-Reply-To: References: <26831043.post@talk.nabble.com> Message-ID: <26831566.post@talk.nabble.com> I only tried once. Maybe someone used to wget could generate a PDF in case people need an offline copy? -- View this message in context: http://old.nabble.com/Mirroring-wiki-with-wget-for-offline-browsing--tp26831043p26831566.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From andrew at hijacked.us Thu Dec 17 09:15:27 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 17 Dec 2009 12:15:27 -0500 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> Message-ID: <20091217171527.GA16380@hijacked.us> On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > Hi, > > I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. > Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. > What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. > > 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? > Tortoise SVN is fine and is probably the de-facto client for windows. > 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? > > 4. There was one fatal error in the build of mod_opal (missing file) > (Some examples of the warnings and the error are shown below :-) > Try with VS 2008 and see if they go away. > 5. How do I specify which options (e.g., mod_flite, to be included iin the build. > You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. > 6. How do I build the sounds etc. ? > The sounds are a subproject too IIRC. Andrew From vizentini at hotmail.com Thu Dec 17 09:18:55 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Thu, 17 Dec 2009 17:18:55 +0000 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl Message-ID: Hi,I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURLBut I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. And sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid Gateway! This is my configuration (overlook language details ) "
"+ ""+ ""+ ""+ ""+ " "+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ "
"+ ""; User id "test" is able to register and call other internal users In my sip_profiles/internal.xml I have: Can you help me with this issue? Thank youPaulo _________________________________________________________________ Keep your friends updated?even when you?re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/b3e1f312/attachment-0002.html From djbinter at yahoo.com Thu Dec 17 09:35:27 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 09:35:27 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> Message-ID: <569367.65001.qm@web37503.mail.mud.yahoo.com> Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/d47e003d/attachment-0002.html From mike at jerris.com Thu Dec 17 09:37:07 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 12:37:07 -0500 Subject: [Freeswitch-users] Polycom 501 conferencing with FreeSwitch In-Reply-To: References: <3C04B27FC880044F8FCD735D0D952FF71780D2D516@EXMBXCLUS01.citservers.local> Message-ID: <7E9B67A4-A2C0-4429-B62C-6B89F2858444@jerris.com> I have not seen anyone mention it. Mike On Dec 17, 2009, at 11:07 AM, Yehavi Bourvine wrote: > I'll rephrase my question: Has anyone done that, or should I dig into it? After all, Polycom is quite common... > > Thanks, __Yehavi: > > 2009/12/17 Michael Jerris > Its software, anything is possible with enough time and effort. > > Mike > > On Dec 17, 2009, at 2:29 AM, Yehavi Bourvine wrote: > > > After some discussions with Polycom support it seems that their conferencing support is based on draft-ietf-sipping-cc-conferencing-03 (which is not the latest and is not compatible with the latest one). > > > > Any idea whether it is possible to program Freeswitch to support this draft? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3a68e92f/attachment-0002.html From brian at freeswitch.org Thu Dec 17 09:44:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 11:44:15 -0600 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: References: Message-ID: <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> I'm going to guess you removed these lines from your profile: parse=true causes the profile to parse the domain looking for gateways and register them.. /b On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote: > Hi, > I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURL > But I am getting this warning: > 2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. > > And > > sofia status gateway MyGW > API CALL [sofia(status gateway MyGW)] output: > Invalid Gateway! > > > This is my configuration (overlook language details ) > > "
"+ > ""+ > ""+ > ""+ > ""+ > " "+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > > ""+ > ""+ > > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > ""+ > "
"+ > ""; > > User id "test" is able to register and call other internal users > > In my sip_profiles/internal.xml I have: > > > > > > > > Can you help me with this issue? > > Thank you > Paulo > > > Keep your friends updated? even when you?re not signed in. _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/edfdb7a9/attachment-0002.html From djbinter at yahoo.com Thu Dec 17 09:47:07 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 09:47:07 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <569367.65001.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <569367.65001.qm@web37503.mail.mud.yahoo.com> Message-ID: <209520.17661.qm@web37504.mail.mud.yahoo.com> I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540 Thank you. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/e78cc4a1/attachment-0002.html From dave at 3c.co.uk Thu Dec 17 09:50:22 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 10:50:22 -0700 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <556038.39248.qm@web37502.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <556038.39248.qm@web37502.mail.mud.yahoo.com> Message-ID: <1261072222.6396.76.camel@local.freepabx.com> Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave > The trace that I pasted on the pastebin was from our > analyzer,Tektronix spectra2 that was sitting between FS and customer. > I also had the FS sip trace on and compare with the trace from Spectra > when I found out about the 3rd re-invite was missing from FS. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 7:57:42 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > The question was: > > Are you doing the packet capture on the actual FS box using tshark or > tcpdump? > > > On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: > We have a customer that we are sending calls to off > the FS and here is the issue: > > > > Call is initially setup fine and they send a first > re-invite with media 0.0.0.0 to place the caller on > hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP > to cut through media and the FS sends a 200 OK to this > fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because > according to them it is strictly for the purpose of > ?keep alive? per RFC 4028. This third re-invite has > the exact same media IP and UDP pot information as the > second re-invite does. The problem is FS does not > respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a > response from FS. > > > One more thing, we did not see the third re-invite in > sofia siptrace, but we do see it in ethereal, which is > kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in > bypass media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn: +19193869900 +19193869900 > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yhding2003 at yahoo.ca Thu Dec 17 09:47:28 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Thu, 17 Dec 2009 09:47:28 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> Message-ID: <26831042.post@talk.nabble.com> Hi, If I configure data as following, why FS A "1001" call FS B "1003" failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26831042.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jerry.richards at teotech.com Thu Dec 17 10:08:37 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 17 Dec 2009 10:08:37 -0800 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: References: Message-ID: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> I found the issue with this. I did an svn checkout from the trunk, and then I did a local svn export to another local folder. For some reason, the svn export did not include the libs/openzap folder (which was not the case when I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap folder? Best Regards, Jerry -----Original Message----- From: Brian West [mailto:brian at freeswitch.org] Sent: Wednesday, December 16, 2009 2:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 Need siptrace with this type "sofia profile xxxx siptrace on" replace xxxx with your profile. /b On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote: > I upgraded to the latest 1.0.5pre9 and now if I try to call from an > internal phone to an external number on my Sangoma PRI, I get a "502 Bad Gateway" > reply. Below is the console loglevel 7 output. It says the > destination is out-of-order. I'm not sure what this means. Any help is appreciated. > > 2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for > proxy > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy > [0] > 2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for > proxy > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy > [0] > 2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 > Rejected by acl "domains". Falling back to Digest auth. > 2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel > sofia/internal/5381 at 192.168.72.141:5060 > [e58e763f-7688-4600-aa70-481bbc359f58] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel > sofia/internal/5381 at 192.168.72.141:5060 entering state [received][100] > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP: > v=0 > o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4 > 192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0 > PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000/1 > a=ptime:20 > a=ptime:20 > > 2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_NEW -> > CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change CS_INIT > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA INIT > 2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_INIT -> > CS_ROUTING > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338 > (sofia/internal/5381 at 192.168.72.141:5060) State INIT going to sleep > 2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA ROUTING > 2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/5381 at 192.168.72.141:5060 Standard ROUTING > 2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing > Anonymous->93491028 in context default > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->unloop] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) > [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) > [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->tod_example] continue=true > Dialplan: day of week[4] =~ 2-6 (PASS) > Dialplan: hour[14] =~ 9-18 (PASS) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match > (PASS) [tod_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(open=true) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->holiday_example] continue=true > Dialplan: month[12] =~ 1 (FAIL) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Date/Time Match > (FAIL) [holiday_example] break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->Mediant1000] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (FAIL) > [Mediant1000] > destination_number(93491028) =~ /^8(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 parsing > [default->SangomaPRI] continue=false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Regex (PASS) > [SangomaPRI] > destination_number(93491028) =~ /^9(\d+)$/ break=on-false > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > set(effective_caller_id_number=425740${caller_id_number}) > Dialplan: sofia/internal/5381 at 192.168.72.141:5060 Action > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_ROUTING -> > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341 > (sofia/internal/5381 at 192.168.72.141:5060) State ROUTING going to sleep > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] mod_sofia.c:181 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA EXECUTE > 2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:159 > sofia/internal/5381 at 192.168.72.141:5060 Standard EXECUTE EXECUTE > sofia/internal/5381 at 192.168.72.141:5060 set(open=true) > 2009-12-16 14:10:46.459538 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET [open]=[true] EXECUTE > sofia/internal/5381 at 192.168.72.141:5060 > set(effective_caller_id_number=4257405381) > 2009-12-16 14:10:46.460549 [DEBUG] mod_dptools.c:768 > sofia/internal/5381 at 192.168.72.141:5060 SET > [effective_caller_id_number]=[4257405381] > EXECUTE sofia/internal/5381 at 192.168.72.141:5060 > bridge(openzap/smg_prid/a/3491028 at g1) > 2009-12-16 14:10:46.479629 [ERR] mod_openzap.c:945 Invalid dial string > 2009-12-16 14:10:46.479629 [ERR] switch_ivr_originate.c:2249 Cannot > create outgoing channel of type [openzap] cause: > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.479629 [DEBUG] switch_ivr_originate.c:3009 > Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [INFO] mod_dptools.c:2303 Originate Failed. > Cause: DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.488521 [NOTICE] mod_dptools.c:2366 Hangup > sofia/internal/5381 at 192.168.72.141:5060 [CS_EXECUTE] > [DESTINATION_OUT_OF_ORDER] > 2009-12-16 14:10:46.488521 [DEBUG] switch_channel.c:1912 Send signal > sofia/internal/5381 at 192.168.72.141:5060 [KILL] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:348 > (sofia/internal/5381 at 192.168.72.141:5060) State EXECUTE going to sleep > 2009-12-16 14:10:46.488521 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:358 Channel > sofia/internal/5381 at 192.168.72.141:5060 hanging up, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] mod_sofia.c:421 Responding to > INVITE > with: 502 > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/5381 at 192.168.72.141:5060 Standard HANGUP, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/5381 at 192.168.72.141:5060) State HANGUP going to sleep > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:333 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_HANGUP -> > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:314 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_REPORTING > 2009-12-16 14:10:46.489603 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/5381 at 192.168.72.141:5060 Standard REPORTING, cause: > DESTINATION_OUT_OF_ORDER > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:585 > (sofia/internal/5381 at 192.168.72.141:5060) State REPORTING going to > sleep > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:327 > (sofia/internal/5381 at 192.168.72.141:5060) State Change CS_REPORTING -> > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1018 Send > signal sofia/internal/5381 at 192.168.72.141:5060 [BREAK] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_session.c:1155 Session > 1 > (sofia/internal/5381 at 192.168.72.141:5060) Locked, Waiting on external > entities > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1173 Session > 1 > (sofia/internal/5381 at 192.168.72.141:5060) Ended > 2009-12-16 14:10:46.562662 [NOTICE] switch_core_session.c:1175 Close > Channel sofia/internal/5381 at 192.168.72.141:5060 [CS_DESTROY] > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/5381 at 192.168.72.141:5060) Running State Change > CS_DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] mod_sofia.c:293 > sofia/internal/5381 at 192.168.72.141:5060 SOFIA DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/5381 at 192.168.72.141:5060 Standard DESTROY > 2009-12-16 14:10:46.562662 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/5381 at 192.168.72.141:5060) State DESTROY going to sleep > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org From mike at jerris.com Thu Dec 17 10:14:28 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 13:14:28 -0500 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <20091217171527.GA16380@hijacked.us> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> Message-ID: On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: >> Hi, >> >> I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? >> >> 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. >> Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. >> What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules stand > a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. >> >> 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? >> > Tortoise SVN is fine and is probably the de-facto client for windows. > make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] >> 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. >> >> 4. There was one fatal error in the build of mod_opal (missing file) >> (Some examples of the warnings and the error are shown below :-) >> > Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. >> 5. How do I specify which options (e.g., mod_flite, to be included iin the build. >> > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer >> 6. How do I build the sounds etc. ? >> > > The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/680983a0/attachment-0002.html From brian at freeswitch.org Thu Dec 17 10:14:31 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 12:14:31 -0600 Subject: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9 In-Reply-To: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> References: <747A23BD2E9049628431027F274E2E1B@greyhawk.tonecommander.com> Message-ID: <8E309ACD-6284-4AE6-BF26-711E45125A5F@freeswitch.org> This would have nothing to do with receiving a 502 on sip. /b On Dec 17, 2009, at 12:08 PM, Jerry Richards wrote: > I found the issue with this. I did an svn checkout from the trunk, and then > I did a local svn export to another local folder. For some reason, the svn > export did not include the libs/openzap folder (which was not the case when > I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap > folder? > > Best Regards, > Jerry From djbinter at yahoo.com Thu Dec 17 10:31:56 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 10:31:56 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <1261072222.6396.76.camel@local.freepabx.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <556038.39248.qm@web37502.mail.mud.yahoo.com> <1261072222.6396.76.camel@local.freepabx.com> Message-ID: <926177.41302.qm@web37504.mail.mud.yahoo.com> Yes, I have a complete trace here: http://pastebin.freeswitch.org/11541 ________________________________ From: David Knell To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:50:22 AM Subject: Re: [Freeswitch-users] SIP Re-invite Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave > The trace that I pasted on the pastebin was from our > analyzer,Tektronix spectra2 that was sitting between FS and customer. > I also had the FS sip trace on and compare with the trace from Spectra > when I found out about the 3rd re-invite was missing from FS. > > Thank you. > > > > ______________________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 7:57:42 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > The question was: > > Are you doing the packet capture on the actual FS box using tshark or > tcpdump? > > > On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: > Anthony, > > I have pasted the invite sip trace here: > http://pastebin.freeswitch.org/11536 > Please advise if you need further info. > > Thank you. > > > > ______________________________________________________________ > From: Anthony Minessale > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 16, 2009 3:42:48 PM > Subject: Re: [Freeswitch-users] SIP Re-invite > > > > that means the invite is not matching the call dialog > compare the via tags and call-id etc > > > On Wed, Dec 16, 2009 at 5:29 PM, DJB > wrote: > We have a customer that we are sending calls to off > the FS and here is the issue: > > > > Call is initially setup fine and they send a first > re-invite with media 0.0.0.0 to place the caller on > hold. FS sends a 200 ok to this first re-invite fine > > > > They then send a second re-invite with their media IP > to cut through media and the FS sends a 200 OK to this > fine. At this point the call is fine > > > > 30 minutes later they send a third re-invite because > according to them it is strictly for the purpose of > ?keep alive? per RFC 4028. This third re-invite has > the exact same media IP and UDP pot information as the > second re-invite does. The problem is FS does not > respond to this third re-invite AT ALL. It doesn?t > send a 100 trying a 200 OK nothing so this causes the > call to be dropped as the other end does not recieve a > response from FS. > > > One more thing, we did not see the third re-invite in > sofia siptrace, but we do see it in ethereal, which is > kind of odds. > > > We are running FreeSWITCH Version 1.0.trunk (15979) in > bypass media mode. > > > Thank you very much. > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn: +19193869900 +19193869900 > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/07c5b91c/attachment-0002.html From yhding2003 at yahoo.ca Thu Dec 17 10:33:09 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Thu, 17 Dec 2009 10:33:09 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <26831042.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> Message-ID: <26832823.post@talk.nabble.com> param name="username" value="1101" param name="password" value="1234" param name="proxy" value="192.168.129.194:5060" param name="register" value="false" Hi, If I configure data as following, why FS A "1001" call FS B "1003" failed ? Thank you! FS A: 192.168.129.168, DN=1001 FS B: 192.168.129.194, DN=1003 In FS A add /conf/sip_proifles/external/gwfsa.xml 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have 1101 number Dan Le wrote: > > If you want FS server A to be able to call FS server B, you can set up a > user account in server B's FS directory configs, and then just treat > server > B as a normal gateway by adding a gateway definition in server A. That > will > allow you to route calls to server B from A; to do the reverse, just > mirror > the configs the other direction. > > On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: > >> >> I like to connect two freeswitch, call each other, communicate and vice >> versa. >> Can you give me an example for that? >> >> Thanks >> -- >> View this message in context: >> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From djbinter at yahoo.com Thu Dec 17 10:53:30 2009 From: djbinter at yahoo.com (DJB) Date: Thu, 17 Dec 2009 10:53:30 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <569367.65001.qm@web37503.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <191c3a030912170757v16951c3bx73dc64af7bc2215f@mail.gmail.com> <569367.65001.qm@web37503.mail.mud.yahoo.com> Message-ID: <25799.44344.qm@web37505.mail.mud.yahoo.com> Please advise whether I should put a request in JIRA. http://pastebin.freeswitch.org/11541 Thank you. ________________________________ From: DJB To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. ________________________________ From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > > > > ________________________________ From: Anthony Minessale >To: freeswitch-users at lists.freeswitch.org >Sent: Wed, December 16, 2009 3:42:48 PM >Subject: Re: [Freeswitch-users] SIP Re-invite > > >>that means the invite is not matching the call dialog >compare the via tags and call-id etc > > > >On Wed, Dec 16, 2009 at 5:29 PM, DJB wrote: > >We have a customer that we are sending calls to off the FS and here is the issue: >> >>Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine >> >>They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine >> >>30 minutes later they send a third re-invite because according to them it is strictly for the purpose of ?keep alive? per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn?t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. >> >> >>One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. >> >> >>We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. >> >> >>Thank you very much. >> >>_______________________________________________ >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >>ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >iax:guest at conference.freeswitch.org/888 >googletalk:conf+888 at conference.freeswitch.org >>pstn:+19193869900 > > >_______________________________________________ >>FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/08f4f658/attachment-0002.html From kristian.kielhofner at gmail.com Thu Dec 17 11:01:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 14:01:16 -0500 Subject: [Freeswitch-users] Handling REFER... Message-ID: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: ------------------------------------------------------------------------ REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: "NONAME" ;tag=BagvZeKSrj7yH From: ;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: Contact: Content-Length: 0 ------------------------------------------------------------------------ send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: ;tag=203332153_1430350929_10 To: "NONAME" ;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From juanbackson at gmail.com Thu Dec 17 11:05:14 2009 From: juanbackson at gmail.com (Juan Backson) Date: Fri, 18 Dec 2009 03:05:14 +0800 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> Message-ID: <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> Hi, I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true and minimum-session-expires=120. Is this the correct way of setting the sip session timers? thanks, jb On Fri, Dec 18, 2009 at 12:13 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > sip session timers is the standardized way to handle this. > > > > On Thu, Dec 17, 2009 at 10:00 AM, Mathieu Rene wrote: > >> Are you doing proxy or bypass meda? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Dec-09, at 12:53 AM, Juan Backson wrote: >> >> Hi >> >> I have rtp-timeout-sec set to 300 s but I am still getting calls with >> duration of 1 day long. >> >> Is there any other ways to check for zombie channels? >> >> jb >> >> On Wed, Dec 16, 2009 at 10:52 PM, Brian West wrote: >> >>> Why not just set rtp-timeout-sec on the sofia profile and it'll do >>> that for you. >>> >>> Unless something else is going on. >>> >>> /b >>> >>> On Dec 16, 2009, at 6:33 AM, Juan Backson wrote: >>> >>> > Hi, >>> > >>> > I am having problem with around 1 % of the channels always get >>> > zombilized. >>> > >>> > What I want to do is to have a background thread that regularly >>> > check all the channels that have been in existance for like > 1 hr, >>> > and then check to see if there is any RTP coming in and going out. >>> > If there is no RTP, then I just hangup that channel. Does anyone >>> > know if there is anyway to do that in a freeswitch module? Which >>> > API can I use to accomplish this purpose? Alternatively, is there >>> > anyway to configure freeswitch so that it will hangup the calls >>> > where there is no media in and out for so many seconds? >>> > >>> > Thanks, >>> > jb >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> > users >>> > http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/eefd9bd7/attachment-0002.html From brian at freeswitch.org Thu Dec 17 11:21:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 13:21:35 -0600 Subject: [Freeswitch-users] detecting rtp packet for zombie channels In-Reply-To: <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> References: <27c25bc40912160433s1e52c1b6sd76210f458f60092@mail.gmail.com> <27c25bc40912162153x1c0c98eh7d60d2cbc81dde5b@mail.gmail.com> <60603A90-5EA2-4434-A2AC-BC0AD958FA0A@avgs.ca> <191c3a030912170813h288a7c1u8ddb5245956d13af@mail.gmail.com> <27c25bc40912171105lc758f73v57b6e7510abd6cb0@mail.gmail.com> Message-ID: Please try on SVN trunk. I might toss a PRE10 sooner. /b On Dec 17, 2009, at 1:05 PM, Juan Backson wrote: > Hi, > > I am using 1.0.4 (exported) with proxy media, and I have enable-timer = true and minimum-session-expires=120. > > Is this the correct way of setting the sip session timers? > > thanks, > jb From brian at proximosystems.com Thu Dec 17 11:29:14 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 14:29:14 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> Message-ID: <049601ca7f4f$37da5580$a78f0080$@com> Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/7f7ba068/attachment-0002.html From brian at freeswitch.org Thu Dec 17 11:33:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 13:33:42 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: If you're going to have that many listeners then it would be best to use something like shoutcast to broadcast the stream out to a local stream on various different boxes... then tie the callers into a stream... when they have questions uuid_transfer them into the conf.. then back to the stream when done. This would scale to very large numbers because you could split it out into 100's of boxes if needed. /b On Dec 17, 2009, at 1:29 PM, Brian wrote: > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I?m doing wrong, but I don?t see what it could be. > > Brian. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/c79a9440/attachment-0002.html From yehavi.bourvine at gmail.com Thu Dec 17 11:36:40 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Dec 2009 21:36:40 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: I am trying Audiocodes and Vegastream ATAs, and work with either the manufacturer or the local representative here. On SNOM I managed to make it work, and will try Polycom soon (once I manage to grab one unit from our users...). Thanks, __yehavi: 2009/12/17 Brian West > Also what device are you using? I haven't tested with many so far... > Polycom, Snom and a few others do TLS (see interop page on wiki) others do > it wrong. > > /b > > On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: > > You could try ssldump: > > http://www.rtfm.com/ssldump/ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/2ea8a523/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 11:42:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 13:42:03 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/4d726120/attachment-0002.html From mike at jerris.com Thu Dec 17 11:43:10 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Dec 2009 14:43:10 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: We are always doing enhancements and yes there are some real scalability enhancements in trunk compared to 1.0.4, I am just not sure if they effect conference significantly or not. I would guess that trunk is actually more stable than 1.0.4 at the moment. Give it a try and find out. Mike On Dec 17, 2009, at 2:29 PM, Brian wrote: > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I?m doing wrong, but I don?t see what it could be. > > Brian. > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > I would be curious what the same tests produce with svn trunk of FreeSWITCH. > > Mike > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn?t scale nearly as well as I?d hoped (based on what I?ve read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I?ve set file logging to ?notice? level. My conference profile is configured to suppress several events, hoping that it would improve performance. > > Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: > > Scenario 1: > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/39907451/attachment-0002.html From steveu at coppice.org Thu Dec 17 11:50:08 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 18 Dec 2009 03:50:08 +0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <049601ca7f4f$37da5580$a78f0080$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> Message-ID: <4B2A8B70.20300@coppice.org> On 12/18/2009 03:29 AM, Brian wrote: > > Hi Mike, > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > Brian. > I don't think you have mentioned which codecs are involved. This can have a profound effect. Steve > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > Mike > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > Here?s my server setup is this: > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the system). > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > Thanks, > > Brian. > > From jeff at jefflenk.com Thu Dec 17 12:06:38 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 17 Dec 2009 12:06:38 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> Message-ID: <1261080398190-4183177.post@n2.nabble.com> The sounds projects (which do the downloads and extraction) are not present for 2005. Also alot of the newer modules dont have build support either. I would suggest you use VS2008 Express Michael Jerris wrote: > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > >> On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: >>> Hi, >>> >>> I'm probably going to regret this - I'm not sure that I'll be able to do >>> this without a lot of pain (nothing to do with FS - more my lack of >>> ability with Visual Studio), but......, I want to try building >>> FreeSwitch from source rather than using the pre-built binaries. I have >>> a couple of initial questions that, hopefully, someone can answer please >>> ? >>> >>> 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on >>> the horizon for me. >>> Having downloaded the SVN, I see there is a VS 2005 Solution, but it is >>> marked as "Unsupported", although the Wiki says that you only need >>> VC++2005. >>> What does "unsupported" mean in this context ? I guess that support for >>> VS2005 is being dropped, but is the VS2005 Solution still being >>> maintained, and if so, for how long? I'd hate to get into the build >>> thing and then find that I was stalled when VS2005 support was dropped >>> altogether ? >> >> Install VS 2008 if at all possible (express edition is free). 2005 >> support isn't maintained much if at all, so a lot of newer modules stand >> a good chance of not having support. > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > >>> >>> 2. The whole SVN thing is new to me but I've worked out that I need an >>> SVN Client on Windows to work with the source. Can anyone recommend the >>> best (free) SVN Client for Windows to use with FreeSwitch. I have >>> installed TortoiseSVN - a Windows Explorer Shell that looks pretty and >>> seemed to work on my first build but it's not command line based so some >>> of the tips given in the Wiki like "make current" and "make sounds" may >>> be more awkward to achieve. Is anyone else using Tortoise and/or can >>> give some tips on which SVN client to use ? >>> >> Tortoise SVN is fine and is probably the de-facto client for windows. >> > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] >>> 3. I built 15979 last night (with VS2005) and got some warnings, with >>> data type conversion - is this a known issue under Windows ? > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > >>> >>> 4. There was one fatal error in the build of mod_opal (missing file) >>> (Some examples of the warnings and the error are shown below :-) >>> >> Try with VS 2008 and see if they go away. > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > >>> 5. How do I specify which options (e.g., mod_flite, to be included iin >>> the build. >>> >> You can enable the different sub projects somehow in the UI, I always >> forget exactly how but just click around in VS and you'll find it. > > You can adjust this in the configuration managaer > >>> 6. How do I build the sounds etc. ? >>> >> >> The sounds are a subproject too IIRC. > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > Mike > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4183177.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Thu Dec 17 12:10:54 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 17 Dec 2009 12:10:54 -0800 (PST) Subject: [Freeswitch-users] [Windows] Stable enough for production use? In-Reply-To: <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> References: <26807322.post@talk.nabble.com> <6E8D2069C08AA84A83D336E996AE4C67032C428586@mse17be1.mse17.exchange.ms> <87f2f3b90912161615k50692451v3e315c2ff6f6246@mail.gmail.com> Message-ID: <1261080654294-4183200.post@n2.nabble.com> I run FreeSWITCH on a Windows Server 2008 R2 (x64) box with several analog lines and it works very well. mercutioviz wrote: > > And we shouldn't be using 1.0.4 anyway, should we? ;) > -MC > > > On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva > wrote: > >> I've been using FreeSWITCH on Windows lately and seems to work pretty >> well. >> Sangoma has been testing more and more lately the Windows drivers with >> FreeSWITCH, and I think you should be just fine.I have not tested 1.0.4 >> though, always using trunk, if you are going to be using PSTN lines (and >> therefore requiring openzap) I think it would be a good idea for you to >> use >> trunk and latest wanpipe drivers. >> >> -- >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R >> 9T3 Canada >> t. 1 905 474 1990 x 128 | e. moy at sangoma.com >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Windows-Stable-enough-for-production-use-tp4174199p4183200.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Dec 17 12:14:58 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 14:14:58 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <4B2A8B70.20300@coppice.org> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <4B2A8B70.20300@coppice.org> Message-ID: Yes, while it is true that does make a profound difference but if he has many listeners and not very many talkers... just tapping into the conference and streaming that audio out would scale well. /b On Dec 17, 2009, at 1:50 PM, Steve Underwood wrote: > I don't think you have mentioned which codecs are involved. This can > have a profound effect. > > Steve From brian at proximosystems.com Thu Dec 17 12:32:23 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 15:32:23 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> Message-ID: <04b501ca7f58$0a188870$1e499950$@com> I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/ae93dca3/attachment-0002.html From vizentini at hotmail.com Thu Dec 17 12:46:00 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Thu, 17 Dec 2009 20:46:00 +0000 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> References: , <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> Message-ID: Hi,FS was sending (while loading modules) such request: [purpose] => gateways But I was not aware of that...so that I am replying FS with my Gateways now... But now I am wondering...suppose I have 1000 domains and two different gateways per domain (2K Gateways) Should I reply FS request with such huge XML on startup? Thanks for your backings PauloFrom: brian at freeswitch.org Date: Thu, 17 Dec 2009 11:44:15 -0600 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl I'm going to guess you removed these lines from your profile: parse=true causes the profile to parse the domain looking for gateways and register them.. /b On Dec 17, 2009, at 11:18 AM, Paulo Vicentini wrote:Hi,I am trying to define Gateways (for inbound and outbound calls via SIP provider) within Directory (under "internal" sample profile) using XML CURLBut I am getting this warning:2009-12-17 14:42:03.294519 [WARNING] sofia_reg.c:2115 Gateway 'MyGW' not found. And sofia status gateway MyGWAPI CALL [sofia(status gateway MyGW)] output:Invalid Gateway! This is my configuration (overlook language details ) "
"+ ""+ ""+ ""+ ""+ " "+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ ""+ "
"+ ""; User id "test" is able to register and call other internal users In my sip_profiles/internal.xml I have: Can you help me with this issue? Thank youPaulo Keep your friends updated? even when you?re not signed in. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/40ac82d6/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 12:48:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 14:48:33 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04b501ca7f58$0a188870$1e499950$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> Message-ID: <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum to use > for this topic from now on. > > > > Thanks, > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 2:42 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > One man's stable release is another man's 6 month old release with hundreds > of known fixed bugs. > If one of the core developers tells you to try it, you may as well take the > time to try it now that you have opened a forum questioning the scalability. > > When you tested asterisk did you actually use 600 phones and verify that > each one can hear the audio perfectly and in time with what the speaker was > saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or follow any > of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have a > policy against entertaining load testing questions but if you like asterisk, > by all means, use it, and good luck to you if those numbers you are testing > at are what you plan to put in real production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/3c69cc8a/attachment-0002.html From dave at 3c.co.uk Thu Dec 17 13:06:48 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 17 Dec 2009 21:06:48 +0000 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04b501ca7f58$0a188870$1e499950$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> Message-ID: <1261084008.6396.89.camel@local.freepabx.com> Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Dec 17 13:07:25 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 15:07:25 -0600 Subject: [Freeswitch-users] [freeswitch-users] Gateway in Directory Loaded via xml curl In-Reply-To: References: , <2B0F61B4-2742-4729-9925-6F110511E801@freeswitch.org> Message-ID: <51C17D71-0193-40BF-9ED6-39A1BA58E5D6@freeswitch.org> In your case don't store them in the domain put them in the gateways tags on the profile directly. /b On Dec 17, 2009, at 2:46 PM, Paulo Vicentini wrote: > Hi, > FS was sending (while loading modules) such request: [purpose] => gateways > But I was not aware of that...so that I am replying FS with my Gateways now... > > But now I am wondering...suppose I have 1000 domains and two different gateways per domain (2K Gateways) > Should I reply FS request with such huge XML on startup? > > > Thanks for your backings > > Paulo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/46de59c3/attachment-0002.html From Prometheus001 at gmx.net Thu Dec 17 13:20:15 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 17 Dec 2009 22:20:15 +0100 Subject: [Freeswitch-users] How to overcome 415 Unsupported Media Type Message-ID: <4B2AA08F.3000102@gmx.net> I try to attach Bravis video conference clients to Freeswitch. Those video conference clients are really working good (Multilingual clients for testing ca be downloaded here: http://www.bravis.eu/). Some big companies here in Germany use them in large installations. They are based on SIP, but do not use any publicly known codecs. Normally they are maintained and routed via our OpenSIPS server, but I would like to integrate them into our Freeswitch system. That way I do not have to manage 2 SIP servers for phone calls and video conferencing calls. However the SIP message does not provide Content-Type: application/sdp. Instead it provides Content-Type: application/BRAVIS. The clients register successfully but they do not invite. Freeswitch answers "SIP/2.0 415 Unsupported Media Type." I have added bypass_media=true into the dialplan and inbound-late-negotiation true in the internal profile but this didn't help. I think Freeswitch complains about the content-type. Is there any way how I may overcome this? Here is a sample Invite INVITE sip:835352 at sip5.mydomain.com SIP/2.0. From: myname ;tag=5c5c3ef6bbe9de119f1aa11f7ca41a5f. To: sip:835352 at sip5.mydomain.com. Via: SIP/2.0/UDP 217.xxx.xxx.xx6:5530;iid=9931;branch=z9hG4bKc4583ef6bbe9de119f1aa11f7ca41a5f;uas-addr=217.24.11.190;rport. CSeq: 4711 INVITE. Call-ID: 2-ee3d3ef6-bbe9-de11-9fa1-a11f7ca41a5f. Contact: "myname" . User-Agent: BRAVIS/1.5.20.27.4585 (Linux 2.6.31-16-generic; generic; Ubuntu 9.10; i686; de; 8). Max-Forwards: 70. Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS. Supported: 100rel. Content-Type: application/BRAVIS. Content-Length: 174. ACAABAAAFDAAABAAMEFBHCGLACAACAAACPKNBHGOAPLDABAAFAAADBABAAPPAAAAELAFAACAAAHDHCGGGMHIPPUPPPPPOPBEKHHHAPLDOPBEKHHHAPLDABAAAAAADCABAAADFBMDHOAEAAAAAAGIGPHDHEAAPPPPPPJFKGAPLHHNKF. Best regards Peter From ujjval at simplesignal.com Thu Dec 17 13:38:38 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Thu, 17 Dec 2009 13:38:38 -0800 Subject: [Freeswitch-users] Performance Tuning Message-ID: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Looking at Performance Tune my Freeswitch http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Is refers to the following": Turn off every module you don't need Turn presence off in the profiles libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles mod_cdr_csv is slower than mod_xml_cdr How do I change each one ....any references on Wiki? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/31ec13fc/attachment-0002.html From brian at proximosystems.com Thu Dec 17 13:41:01 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 16:41:01 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> Message-ID: <04ca01ca7f61$a0560e30$e1022a90$@com> I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 3:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability We didn't post it anywhere but we just get overwhelmed with them and many of them are unfounded and take up a lot of time to track down. That does not mean you have not found a real problem but the first step is trying trunk. On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: I didn't realize there was a policy about load testing questions. What forum should I have used for this? I didn't get the chance to test on FS trunk yet, but when I do I will provide you with the feedback when I do. Just let me know what forum to use for this topic from now on. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, December 17, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability One man's stable release is another man's 6 month old release with hundreds of known fixed bugs. If one of the core developers tells you to try it, you may as well take the time to try it now that you have opened a forum questioning the scalability. When you tested asterisk did you actually use 600 phones and verify that each one can hear the audio perfectly and in time with what the speaker was saying? Did you try same on FS? Did you optimize your dialplan on FS to deal with a load test or follow any of the recommended performance tuning page. All of the answers to these questions are really moot because we have a policy against entertaining load testing questions but if you like asterisk, by all means, use it, and good luck to you if those numbers you are testing at are what you plan to put in real production......... On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: Hi Mike, I didn't get around to testing on the FreeSWITCH trunk yet. Are there substantial fixes to mod_conference in the FreeSWITCH trunk that might increase capacity for my scenario of one speaker and many listeners? If I want to put this into a production environment, I would need a stable version, which as far as I know is the 1.0.4 version. However, I did test on Asterisk 1.4 using app_conference, and doing the same scenario was able to get 1 speaker and 600 listeners on a single conference with no audio issues. The CPU at that point was just over 300%, same as where the single conference scenario failed on FreeSWITCH with 300 listeners. I was able to push it to over 700 listeners before I reached 400% CPU usage (I guess maxing out my quad-core processors), and asterisk finally crashed. But up until that point, there were no audio problems. I've read a lot about how FreeSWITCH is supposed to be more scalable than Asterisk, but unless there is something wrong with my FreeSWITCH setup, Asterisk was clearly the winner in this test - more than doubling FreeSWITCH capacity in this case. Again, maybe there is something on the FreeSWITCH side that I'm doing wrong, but I don't see what it could be. Brian. From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, December 17, 2009 10:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I would be curious what the same tests produce with svn trunk of FreeSWITCH. Mike On Dec 16, 2009, at 4:49 PM, Brian wrote: Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/5b0fa004/attachment-0002.html From vinuth.madinur at gmail.com Thu Dec 17 13:53:55 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Fri, 18 Dec 2009 03:23:55 +0530 Subject: [Freeswitch-users] Performance Tuning In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Message-ID: <910309030912171353v17372138idd3c8e1cb5b15ee1@mail.gmail.com> 1. http://wiki.freeswitch.org/wiki/Modules.conf.xml 2. http://wiki.freeswitch.org/wiki/Sofia.conf.xml#manage-presence 3. http://wiki.freeswitch.org/wiki/Getting_Started_Guide#SIP_Profiles Might not be entirely helpful, but basically you can use either the external or internal profiles and change the ports, etc., as required. 4. You can disable mod_cdr_csv and enable mod_xml_cdr based on #1. Thanks, Vinuth. On Fri, Dec 18, 2009 at 3:08 AM, Ujjval Karihaloo wrote: > Looking at Performance Tune my Freeswitch > > > > http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations > > > > > > > > Is refers to the following?: > > > > > > Turn off every module you don't need > > Turn presence off in the profiles > > libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles > > mod_cdr_csv is slower than mod_xml_cdr > > > > > > How do I change each one ?.any references on Wiki? > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/f710c9ed/attachment-0002.html From brian at proximosystems.com Thu Dec 17 14:05:24 2009 From: brian at proximosystems.com (Brian) Date: Thu, 17 Dec 2009 17:05:24 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261084008.6396.89.camel@local.freepabx.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <1261084008.6396.89.camel@local.freepabx.com> Message-ID: <04e901ca7f65$08edd830$1ac98890$@com> Hi Dave, That was one of the questions I had in my original post, was there an alternative way to implement a single speaker, many listener case? There was a suggestion proposed to use local streams instead of the conference. I'm not familiar with it, and I'm in the process of reading the wiki and source code to see what can be done with that. Thanks, Brian. -----Original Message----- From: David Knell [mailto:dave at 3c.co.uk] Sent: Thursday, December 17, 2009 4:07 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker. As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 17 14:06:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:06:04 -0600 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> Message-ID: <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> The calls inherit the context from the parent, I think there is a var you can set on the chan to pick what context to use in a transfer like transfer_context or something grep the code for it On Dec 17, 2009 1:07 PM, "Kristian Kielhofner" < kristian.kielhofner at gmail.com> wrote: Hello everyone, I've got two profiles running: s2s and trunk. The context for s2s is defined as s2s-in. The context for trunk is defined as trunk-in. trunk is bound to 192.168.168.3. recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: ------------------------------------------------------------------------ REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 To: "NONAME" >;tag=BagvZeKSrj7yH From: ;tag=203332153_1430350929_10 Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Max-Forwards: 70 Refer-To: > Contact: Content-Length: 0 ------------------------------------------------------------------------ send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 From: ;tag=203332153_1430350929_10 To: "NONAME" >;tag=BagvZeKSrj7yH Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac CSeq: 2 REFER Contact: User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 FS routed this to the s2s-in context, even though it was sent to the trunk profile. Shouldn't it have ended up in trunk-in? For the time being I wrote some crazy dialplan for s2s-in to transfer the call to trunk-in but I'm wondering what could be going on here. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dc6db1a8/attachment-0002.html From timuckun at gmail.com Thu Dec 17 14:13:33 2009 From: timuckun at gmail.com (Tim Uckun) Date: Fri, 18 Dec 2009 11:13:33 +1300 Subject: [Freeswitch-users] Performance Tuning In-Reply-To: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> References: <3C04B27FC880044F8FCD735D0D952FF717FA4EFA83@EXMBXCLUS01.citservers.local> Message-ID: <855e4dcf0912171413s36ed4453m1be00e454f89562a@mail.gmail.com> > > libsofia only handles 1 thread per profile, so if that is your bottle neck > use more profiles If you only have one provider for your trunk is it possible to set up multiple profiles for enhanced performance? For example if I have multiple DDIs from the provider can I set up a different profile for each one? Or maybe based on some some sort of a pattern? From Prometheus001 at gmx.net Thu Dec 17 14:17:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 17 Dec 2009 23:17:00 +0100 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <010101ca7f33$d9359eb0$8ba0dc10$@de> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> Message-ID: <4B2AADDC.1030805@gmx.net> Hello Oliver, I have the same on Ubuntu wth newest trunk. Best regards Peter Oliver Sch?nbeck schrieb: > > Hello, > > > > we are running freeswitch 1.0.trunk and are currently trying to get > the mod_voicemail to send the received messages to the user by using > exim4 on a debian machine. > > > > So far we followed the instructions in the wiki article ( > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > > typeset LOG="/tmp/${0##*/}.out" > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > exec > $LOG 2>&1 > > exim4 -t -v >> $LOG > > > > If I run the script from the command line everything is working as > expected. If the script gets called by freeswitch I get the following > result in my logfile: > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > fault (core dumped) exim4 -t -v >> $LOG > > > > Has anybody seen similar effects before? > > > > Any advice whats going wrong is heavily appreciated. > > > > Thanks > > Oliver > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Thu Dec 17 14:20:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 16:20:24 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: > I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. > > Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/47dfc698/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 14:46:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:46:03 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp On Thu, Dec 17, 2009 at 3:41 PM, Brian wrote: > I did a test with the trunk version for the one conference case, and it > is the same results as for 1.0.4. The audio failed at around 300 listeners. > Oddly though, it consumed less %CPU (240% instead of 300%), and yet the > audio still failed at the same number of listeners. > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 3:49 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > We didn't post it anywhere but we just get overwhelmed with them and many > of them are unfounded and take up a lot of time to track down. That does > not mean you have not found a real problem but the first step is trying > trunk. > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum to use > for this topic from now on. > > > > Thanks, > > > > Brian. > > > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, December 17, 2009 2:42 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > One man's stable release is another man's 6 month old release with hundreds > of known fixed bugs. > If one of the core developers tells you to try it, you may as well take the > time to try it now that you have opened a forum questioning the scalability. > > When you tested asterisk did you actually use 600 phones and verify that > each one can hear the audio perfectly and in time with what the speaker was > saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or follow any > of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have a > policy against entertaining load testing questions but if you like asterisk, > by all means, use it, and good luck to you if those numbers you are testing > at are what you plan to put in real production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? If I > want to put this into a production environment, I would need a stable > version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing the > same scenario was able to get 1 speaker and 600 listeners on a single > conference with no audio issues. The CPU at that point was just over 300%, > same as where the single conference scenario failed on FreeSWITCH with 300 > listeners. I was able to push it to over 700 listeners before I reached > 400% CPU usage (I guess maxing out my quad-core processors), and asterisk > finally crashed. But up until that point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable than > Asterisk, but unless there is something wrong with my FreeSWITCH setup, > Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH > capacity in this case. Again, maybe there is something on the FreeSWITCH > side that I?m doing wrong, but I don?t see what it could be. > > > > Brian. > > > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > Mike > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > Hi, > > > > I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to > see if it will scale better that other solutions. My scenario is to have one > speaker, and many listeners (mute). Since I have only one speaker, I was > expecting this to scale well because there is no audio mixing required, just > send each frame of the single speaker to each listener. Unfortunately, my > testing was disappointing, and it didn?t scale nearly as well as I?d hoped > (based on what I?ve read on how FreeSWITCH is supposed to be generally very > scalable). > > > > Here?s my server setup is this: > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of > RAM. I?ve set file logging to ?notice? level. My conference profile is > configured to suppress several events, hoping that it would improve > performance. > > > > Here are a few scenarios I tested, and roughly where I reached the point of > audio failure on the conferences: > > > > Scenario 1: > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > Scenario 2: > > 4 conferences, 1 speaker per conference, audio failed approx 110 listeners > per conference (so just over 400 total channels on the system). > > > > Scenario 3: > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners per > conference (so just over 500 total channels on the system). > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, the > audio quality failed when the % CPU for the FreeSWITCH process exceeded > 300%. > > > > I was hoping maybe someone else might have done similar testing, or maybe > has suggestions on how to improve the performance. Or perhaps an alternate > solution to the one speaker, many listener case? > > > > Thanks, > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/587668b6/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 17 14:47:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Dec 2009 16:47:01 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: <4B2AADDC.1030805@gmx.net> References: <010101ca7f33$d9359eb0$8ba0dc10$@de> <4B2AADDC.1030805@gmx.net> Message-ID: <191c3a030912171447o16c81a8bp34886d8ce4d923c5@mail.gmail.com> yah it's exim segfaulting because you have to configure it to emulate sendmail per the wiki page. On Thu, Dec 17, 2009 at 4:17 PM, Peter P GMX wrote: > Hello Oliver, > > I have the same on Ubuntu wth newest trunk. > > Best regards > Peter > > Oliver Sch?nbeck schrieb: > > > > Hello, > > > > > > > > we are running freeswitch 1.0.trunk and are currently trying to get > > the mod_voicemail to send the received messages to the user by using > > exim4 on a debian machine. > > > > > > > > So far we followed the instructions in the wiki article ( > > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > > > > > I added some lines to the bash script to enable some kind of logging: > > #! /bin/bash > > > > typeset LOG="/tmp/${0##*/}.out" > > > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > > > exec > $LOG 2>&1 > > > > exim4 -t -v >> $LOG > > > > > > > > If I run the script from the command line everything is working as > > expected. If the script gets called by freeswitch I get the following > > result in my logfile: > > > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > > fault (core dumped) exim4 -t -v >> $LOG > > > > > > > > Has anybody seen similar effects before? > > > > > > > > Any advice whats going wrong is heavily appreciated. > > > > > > > > Thanks > > > > Oliver > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/22c65448/attachment-0002.html From kristian.kielhofner at gmail.com Thu Dec 17 15:59:45 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 18:59:45 -0500 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> Message-ID: <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> Thanks for the hint! force_transfer_context and force_transfer_dialplan. I've updated the wiki (I'll add an example once I test it). On Thu, Dec 17, 2009 at 5:06 PM, Anthony Minessale wrote: > The calls inherit the context from the parent, I think there is a var you > can set on the chan to pick what context to use in a transfer like > transfer_context or something grep the code for it > > On Dec 17, 2009 1:07 PM, "Kristian Kielhofner" > wrote: > > Hello everyone, > > I've got two profiles running: s2s and trunk. ?The context for s2s is > defined as s2s-in. ?The context for trunk is defined as trunk-in. > trunk is bound to 192.168.168.3. > > recv 481 bytes from udp/[192.168.168.76]:5065 at 18:43:37.309706: > ? ------------------------------------------------------------------------ > ? REFER sip:mod_sofia at 192.168.168.3:5060 SIP/2.0 > ? Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 > ? To: "NONAME" ;tag=BagvZeKSrj7yH > ? From: > ;tag=203332153_1430350929_10 > ? Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac > ? CSeq: 2 REFER > ? Max-Forwards: 70 > ? Refer-To: > ? Contact: > ? Content-Length: 0 > > ? ------------------------------------------------------------------------ > send 592 bytes to udp/[192.168.168.76]:5065 at 18:43:37.316093: > ? ------------------------------------------------------------------------ > ? SIP/2.0 202 Accepted > ? Via: SIP/2.0/UDP 192.168.168.76:5065;branch=z9hG4bK__7539073431157335561_9 > ? From: > ;tag=203332153_1430350929_10 > ? To: "NONAME" ;tag=BagvZeKSrj7yH > ? Call-ID: e505f332-65de-122d-d183-eb12ad0ec1ac > ? CSeq: 2 REFER > ? Contact: > ? User-Agent: FreeSWITCH > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, > SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > ? Supported: precondition, path, replaces > ? Allow-Events: talk, refer > ? Content-Length: 0 > > ?FS routed this to the s2s-in context, even though it was sent to the > trunk profile. ?Shouldn't it have ended up in trunk-in? ?For the time > being I wrote some crazy dialplan for s2s-in to transfer the call to > trunk-in but I'm wondering what could be going on here. > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From frank at impactfax.com Thu Dec 17 16:01:50 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 17 Dec 2009 19:01:50 -0500 Subject: [Freeswitch-users] sip message logging and analysis Message-ID: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> I bit off topic but. Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/feb001e5/attachment-0002.html From freeswitch at aastral.net Thu Dec 17 16:01:35 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 19:01:35 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> Message-ID: <4B2AC65F.5090806@aastral.net> Hey Brian, I've been doing some testing and I am unable to get auth-calls to work through a proxy the way I want them to, even with setting apply-proxy-acl to either the endpoint IP or the proxy IP. I have a multi-tenant system with multiple domains with multiple users in each domain. And I want to restrict a user to an arbitrary CIDR and challenge them for a password. The arbitrary CIDR will vary from UA to UA, and is specified in the directory via the auth-acl parameter. TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of the proxy. Thanks, Bill Brian West wrote: > it needs to be an ACL from acl.conf or a ip/cidr > > /b > > On Dec 17, 2009, at 5:41 AM, Bill W wrote: > >> Okay, I added: to my sofia >> profile and restarted sofia, and still no joy. >> >> I'm on FreeSWITCH Version 1.0.trunk (15764) >> I've got in >> the directory, but I'm still being rejected by the acl: >> >> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >> Rejected by user acl 190.218.103.12/32 >> >> Here's what I believe is the appropriate snippet of the debug output: >> http://pastebin.freeswitch.org/11531 >> >> Thoughts? >> Thanks, >> Bill > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kristian.kielhofner at gmail.com Thu Dec 17 16:27:16 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Thu, 17 Dec 2009 19:27:16 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> Frank, Probably the cleanest (albeit non-FreeSWITCH) way to implement this would be to use OpenSIPS/SER/etc between you and the carrier with the siptrace module. But that's probably more work than you want. There's always tcpdump with a decent filter (udp port 5060 and host x.x.x.x) and then something like http://www.badpenguin.co.uk/files/pcap-util2 Both will allow you to search for BYEs and who is sending them. Also keep in mind that they (or you) may just be dropping the RTP without ever sending a BYE. Setting the various RTP timeouts in FreeSWITCH can help with that. You can then look for logs/events (are there any for RTP timeout?) to see who's dropping RTP. On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact wrote: > I bit off topic but? > > > > Using FS to send calls sip to the LD carrier. > > > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call.? But this is? a > day later after the trouble has been reported. > > > > I am looking for guidance on how to log all sip message traffic and then be > able to easily retrieve to find a call and look at what sip messages really > were being based and by whom.? Maybe store them in a database or some other > file that might be opened by an analysis tool. > > > > Any suggestions on how to log this information and then what tool to use for > later analysis? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Thu Dec 17 16:27:58 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Dec 2009 16:27:58 -0800 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> Message-ID: <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> On Thu, Dec 17, 2009 at 3:59 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Thanks for the hint! > > force_transfer_context and force_transfer_dialplan. > > I've updated the wiki (I'll add an example once I test it). > > I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dba64367/attachment-0002.html From msc at freeswitch.org Thu Dec 17 16:33:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Dec 2009 16:33:11 -0800 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <87f2f3b90912171633v4ae885afv873cba3aa9cde23d@mail.gmail.com> On Thu, Dec 17, 2009 at 4:01 PM, Frank @ Impact wrote: > I bit off topic but? > > > > Using FS to send calls sip to the LD carrier. > > > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call. But this is aday later after the trouble has been reported. > > > > I am looking for guidance on how to log all sip message traffic and then be > able to easily retrieve to find a call and look at what sip messages really > were being based and by whom. Maybe store them in a database or some > other file that might be opened by an analysis tool. > > > > Any suggestions on how to log this information and then what tool to use > for later analysis? > > > Jason Garland's ClueCon2009 videos about tcpdump and wireshark cover the thought of doing a rotating log file so that it captures a bunch of stuff but doesn't go over X number of megabytes... I don't recall exactly where in his videos that part appears, but here are the links to those vids. Hope it helps! -MC Look at this video first: http://www.viddler.com/explore/cluecon/videos/33/ Then check this one if you need more info: http://www.viddler.com/explore/cluecon/videos/8/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/dc372834/attachment-0002.html From chris at fowler.cc Thu Dec 17 16:33:34 2009 From: chris at fowler.cc (Chris Fowler) Date: Thu, 17 Dec 2009 19:33:34 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> I'm using VQManager (there is a 30 day trial) and it's useful for seeing who does what / when per call; it's very easy to install... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Frank @ Impact Sent: Thursday, December 17, 2009 4:02 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] sip message logging and analysis I bit off topic but... Using FS to send calls sip to the LD carrier. Some calls have problems where they drop the call or audio drops or whatever. The carrier's first response is that we dropped the call. But this is a day later after the trouble has been reported. I am looking for guidance on how to log all sip message traffic and then be able to easily retrieve to find a call and look at what sip messages really were being based and by whom. Maybe store them in a database or some other file that might be opened by an analysis tool. Any suggestions on how to log this information and then what tool to use for later analysis? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/82660632/attachment-0002.html From brian at freeswitch.org Thu Dec 17 16:54:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 18:54:04 -0600 Subject: [Freeswitch-users] Handling REFER... In-Reply-To: <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> References: <2d9149cd0912171101k302b43m181b3510316a98c6@mail.gmail.com> <191c3a030912171406s6a6a1f12s57c71e8ea6f52a0d@mail.gmail.com> <2d9149cd0912171559m4360871fl82504efb957bd9a8@mail.gmail.com> <87f2f3b90912171627q4961cf57h47ee3fbda3552f60@mail.gmail.com> Message-ID: <99531772-685B-4CB0-AA48-E2F023A35366@freeswitch.org> Also when can we expect little KK's running around? :P Congrats on the marriage!!!! /b On Dec 17, 2009, at 6:27 PM, Michael Collins wrote: > I love it when users go all Chuck Norris and Rambo in answering their questions AND documenting the info! Thanks KK. > > -MC From brian at freeswitch.org Thu Dec 17 16:54:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Dec 2009 18:54:44 -0600 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <7454A296C7EDE34EA57199FAA401E2F11B7E11A748@VMBX113.ihostexchange.net> Message-ID: <3CA5BAB4-A966-41F9-BAB6-4C4ED91CBA03@freeswitch.org> So is wireshark UI and its free! :P /b On Dec 17, 2009, at 6:33 PM, Chris Fowler wrote: > I?m using VQManager (there is a 30 day trial) and it?s useful for seeing who does what / when per call; it?s very easy to install? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/64fe187d/attachment-0002.html From david.villasmil.work at gmail.com Thu Dec 17 16:54:49 2009 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 18 Dec 2009 01:54:49 +0100 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> Message-ID: <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> i agree with christian, though i would use tshark. you can actually get the fields you want (method and callid) and store them in a dB. then you need to match them with a query. it is simple but Lots of work. look into -e and -E of tshark separate the fields by "," have fun! David El 18/12/2009, a las 01:27, Kristian Kielhofner escribi?: > Frank, > > Probably the cleanest (albeit non-FreeSWITCH) way to implement this > would be to use OpenSIPS/SER/etc between you and the carrier with the > siptrace module. > > But that's probably more work than you want. There's always tcpdump > with a decent filter (udp port 5060 and host x.x.x.x) and then > something like http://www.badpenguin.co.uk/files/pcap-util2 > > Both will allow you to search for BYEs and who is sending them. > > Also keep in mind that they (or you) may just be dropping the RTP > without ever sending a BYE. Setting the various RTP timeouts in > FreeSWITCH can help with that. You can then look for logs/events (are > there any for RTP timeout?) to see who's dropping RTP. > > On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact > wrote: >> I bit off topic but? >> >> >> >> Using FS to send calls sip to the LD carrier. >> >> >> >> Some calls have problems where they drop the call or audio drops or >> whatever. >> >> The carrier?s first response is that we dropped the call. But thi >> s is a >> day later after the trouble has been reported. >> >> >> >> I am looking for guidance on how to log all sip message traffic and >> then be >> able to easily retrieve to find a call and look at what sip >> messages really >> were being based and by whom. Maybe store them in a database or >> some other >> file that might be opened by an analysis tool. >> >> >> >> Any suggestions on how to log this information and then what tool >> to use for >> later analysis? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dujinfang at gmail.com Thu Dec 17 17:21:25 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 18 Dec 2009 09:21:25 +0800 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> <2d9149cd0912171627p1a0bf6cm110557916a38f174@mail.gmail.com> <0A9287F5-021D-45FC-9A1A-9552FAE38A09@gmail.com> Message-ID: <23f91030912171721o6d4b9917ree41ec3092fb8a96@mail.gmail.com> I'm using contrib/seven/sip/sip2db.rb 2009/12/18 David Villasmil : > i agree with christian, though i would use tshark. you can actually > get the fields you want (method and callid) and store them in a dB. > then you need to match them with a query. it is simple but Lots of work. > > look into -e and -E of tshark separate the fields by "," > > have fun! > > David > > El 18/12/2009, a las 01:27, Kristian Kielhofner ?> escribi?: > >> Frank, >> >> ?Probably the cleanest (albeit non-FreeSWITCH) way to implement this >> would be to use OpenSIPS/SER/etc between you and the carrier with the >> siptrace module. >> >> ?But that's probably more work than you want. ?There's always tcpdump >> with a decent filter (udp port 5060 and host x.x.x.x) and then >> something like http://www.badpenguin.co.uk/files/pcap-util2 >> >> ?Both will allow you to search for BYEs and who is sending them. >> >> ?Also keep in mind that they (or you) may just be dropping the RTP >> without ever sending a BYE. ?Setting the various RTP timeouts in >> FreeSWITCH can help with that. ?You can then look for logs/events (are >> there any for RTP timeout?) to see who's dropping RTP. >> >> On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact >> wrote: >>> I bit off topic but? >>> >>> >>> >>> Using FS to send calls sip to the LD carrier. >>> >>> >>> >>> Some calls have problems where they drop the call or audio drops or >>> whatever. >>> >>> The carrier?s first response is that we dropped the call. ?But thi >>> s is ?a >>> day later after the trouble has been reported. >>> >>> >>> >>> I am looking for guidance on how to log all sip message traffic and >>> then be >>> able to easily retrieve to find a call and look at what sip >>> messages really >>> were being based and by whom. ?Maybe store them in a database or >>> some other >>> file that might be opened by an analysis tool. >>> >>> >>> >>> Any suggestions on how to log this information and then what tool >>> to use for >>> later analysis? >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> Kristian Kielhofner >> http://www.astlinux.org >> http://blog.krisk.org >> http://www.star2star.com >> http://www.submityoursip.com >> http://www.voalte.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Thu Dec 17 17:36:50 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 18 Dec 2009 09:36:50 +0800 Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <26832823.post@talk.nabble.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> <26832823.post@talk.nabble.com> Message-ID: <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> I couldn't guess what you want, pastbin your full config and logs and give more detail of your story perhaps someone can help you. 2009/12/18 yvonne ding : > > param name="username" value="1101" > param name="password" value="1234" > param name="proxy" value="192.168.129.194:5060" > param name="register" value="false" > > > Hi, > > If I configure data as following, why FS A "1001" call FS B "1003" failed ? > Thank you! > > FS A: 192.168.129.168, DN=1001 > FS B: 192.168.129.194, DN=1003 > > In FS A add /conf/sip_proifles/external/gwfsa.xml > ? > ? ? > > > > > ? ? > > > 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have > 1101 number > > > > > > Dan Le wrote: >> >> If you want FS server A to be able to call FS server B, you can set up a >> user account in server B's FS directory configs, and then just treat >> server >> B as a normal gateway by adding a gateway definition in server A. That >> will >> allow you to route calls to server B from A; to do the reverse, just >> mirror >> the configs the other direction. >> >> On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz wrote: >> >>> >>> I like to connect two freeswitch, call each other, communicate and vice >>> versa. >>> Can you give me an example for that? >>> >>> Thanks >>> -- >>> View this message in context: >>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-users-list at metik.com Thu Dec 17 18:43:28 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 21:43:28 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AC65F.5090806@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> Message-ID: <4B2AEC50.3030305@metik.com> This may be difficult considering that ACL needs to consider the original src IP/URI. To do that it, freeswitch would need to do so using a header that retains that information (i.e. From, Via, Contact, etc.). Which I do not believe is currently possible using auth-acl or apply-proxy-acl. However, you should be able to emulate the behavior using mod_xml_curl (and validating against appropriate variables available when using it to authenticate the request). see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization -metik Bill W wrote: > Hey Brian, > > > I've been doing some testing and I am unable to get auth-calls to work > through a proxy the way I want them to, even with setting > apply-proxy-acl to either the endpoint IP or the proxy IP. > > I have a multi-tenant system with multiple domains with multiple users > in each domain. And I want to restrict a user to an arbitrary CIDR and > challenge them for a password. The arbitrary CIDR will vary from UA to > UA, and is specified in the directory via the auth-acl parameter. > > TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of > the proxy. > > > Thanks, > Bill > > Brian West wrote: > >> it needs to be an ACL from acl.conf or a ip/cidr >> >> /b >> >> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >> >> >>> Okay, I added: to my sofia >>> profile and restarted sofia, and still no joy. >>> >>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>> I've got in >>> the directory, but I'm still being rejected by the acl: >>> >>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>> Rejected by user acl 190.218.103.12/32 >>> >>> Here's what I believe is the appropriate snippet of the debug output: >>> http://pastebin.freeswitch.org/11531 >>> >>> Thoughts? >>> Thanks, >>> Bill >>> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Thu Dec 17 18:56:26 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 18:56:26 -0800 (PST) Subject: [Freeswitch-users] Creating Default Accounts on Directory Message-ID: <26838457.post@talk.nabble.com> Hi Sir, I want to create a new xml file on the default directory of freeswitch where 1000.xml is located, sample i created 9387821.xml and copy the contents of the 1000.xml. The problem is when I used the account 9387821.xml and call 1000.xml it doesn't work the message in freeswitch it always CS_DESTROY... Please help me this with issue thanks... Edmar -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at freeswitch.org Thu Dec 17 19:08:03 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 18 Dec 2009 01:08:03 -0200 Subject: [Freeswitch-users] Creating Default Accounts on Directory In-Reply-To: <26838457.post@talk.nabble.com> References: <26838457.post@talk.nabble.com> Message-ID: Please check your dialplan to match the new extension. You are looking for dialplan/default.xml extension Local_Extension. Check the cond destination_number, it should give you a good hint. Regards, JM On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz wrote: > > Hi Sir, > > I want to create a new xml file on the default directory of freeswitch > where 1000.xml is located, sample i created 9387821.xml and copy the > contents of the 1000.xml. > > The problem is when I used the account 9387821.xml and call 1000.xml it > doesn't work the message in freeswitch it always CS_DESTROY... Please help > me this with issue thanks... > > Edmar > > > -- > View this message in context: > http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/722a566d/attachment-0002.html From freeswitch-users-list at metik.com Thu Dec 17 19:42:14 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 22:42:14 -0500 Subject: [Freeswitch-users] sip message logging and analysis In-Reply-To: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> References: <063E5F01BA5C41BF8DF4666B640C3FAE@ws4> Message-ID: <4B2AFA16.8020200@metik.com> Some providers do retain call data for diagnostic purposes and to to aid in troubleshooting. Why not politely ask them if they could provide you with a sip trace themselves or forward along the evidence that supported their conclusion. They should be willing to help you solve a problem that may potentially be of benefit to their other customers that report similar issues. Otherwise, as others suggest, you could simply capture the signaling and media traffic from the FS box itself using "tcpdump" (e.g. tcpdump -i eth0 -s 0 -w debug.pcap host 127.0.0.1 ) or ngrep (-d eth0 -W byline -O /tmp/debug.pcap host 127.0.0.1) and analyze the resulting file in Wirehark (Statistics->Voip Calls or Telephony->Voip Calls in the current version). If your provider is using a session border controller or does not have a distributed architecture, then you can replace 127.0.0.1 with the appropriate address. If not, then simply don't use the host filter at all (it will result in a larger capture file). I would just keep in mind that if an upstream device (NAT router, firewall, etc.) is wreaking havoc with session refreshes by dropping re-INVITEs or UPDATEs (associated with session refreshing), you may not see them because of your vantage point. The reason I typically recommend using the "-i" (tcpdump) and "-d" (ngrep) switch is to avoid linux 'cooked' captures (more of a personal preference since I occasionally do have to convert or merge captures). If you only have SSH access to your FS box, you may want to use tcpdump or ngrep along with "screen". "tshark" (tty/cli vesion of Wireshark) and "sipgrep" are also extremely useful. The later requires ngrep and a couple perl modules but I believe it is included with FS in the contrib or scripts directory--I forget which). -metik Frank @ Impact wrote: > > I bit off topic but? > > Using FS to send calls sip to the LD carrier. > > Some calls have problems where they drop the call or audio drops or > whatever. > > The carrier?s first response is that we dropped the call. But this is > a day later after the trouble has been reported. > > I am looking for guidance on how to log all sip message traffic and > then be able to easily retrieve to find a call and look at what sip > messages really were being based and by whom. Maybe store them in a > database or some other file that might be opened by an analysis tool. > > Any suggestions on how to log this information and then what tool to > use for later analysis? > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From darklion11 at yahoo.com Thu Dec 17 19:53:23 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 19:53:23 -0800 (PST) Subject: [Freeswitch-users] Creating Default Accounts on Directory In-Reply-To: References: <26838457.post@talk.nabble.com> Message-ID: <26838750.post@talk.nabble.com> Hi Sir, Not working i set this to to call 8000001.xml up to 8000009.xml on the dialplan/default.xml same thing... Thanks, Edmar Jo?o Mesquita-4 wrote: > > Please check your dialplan to match the new extension. > > You are looking for dialplan/default.xml extension Local_Extension. Check > the cond destination_number, it should give you a good hint. > > Regards, > > JM > > On Fri, Dec 18, 2009 at 12:56 AM, Edmar Cruz wrote: > >> >> Hi Sir, >> >> I want to create a new xml file on the default directory of >> freeswitch >> where 1000.xml is located, sample i created 9387821.xml and copy the >> contents of the 1000.xml. >> >> The problem is when I used the account 9387821.xml and call 1000.xml >> it >> doesn't work the message in freeswitch it always CS_DESTROY... Please >> help >> me this with issue thanks... >> >> Edmar >> >> >> -- >> View this message in context: >> http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838457.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Creating-Default-Accounts-on-Directory-tp26838457p26838750.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 17 20:02:48 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 17 Dec 2009 23:02:48 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AEC50.3030305@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> Message-ID: <4B2AFEE8.5020002@aastral.net> Hey Metik, Thanks for the reply, and the pointers for doing it with xml_curl. I'll guess have to do that in the short term, but in my opinion, having auth-acl be able to work through a proxy is very important as it is a vital part of a comprehensive security feature set. And it would be much simpler to implement from an end-user perspective than the alternative of doing it in xml_curl. As a matter of fact, I'm considering offering a bounty for that feature. What is the going rate for that kind of thing? Is anyone out there interested in coding this feature? Or chipping in for the bounty? Thanks, Bill Metik wrote: > This may be difficult considering that ACL needs to consider the > original src IP/URI. To do that it, freeswitch would need to do so > using a header that retains that information (i.e. From, Via, Contact, > etc.). Which I do not believe is currently possible using auth-acl or > apply-proxy-acl. > > However, you should be able to emulate the behavior using mod_xml_curl > (and validating against appropriate variables available when using it to > authenticate the request). > > see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization > > -metik > > > Bill W wrote: >> Hey Brian, >> >> >> I've been doing some testing and I am unable to get auth-calls to work >> through a proxy the way I want them to, even with setting >> apply-proxy-acl to either the endpoint IP or the proxy IP. >> >> I have a multi-tenant system with multiple domains with multiple users >> in each domain. And I want to restrict a user to an arbitrary CIDR and >> challenge them for a password. The arbitrary CIDR will vary from UA to >> UA, and is specified in the directory via the auth-acl parameter. >> >> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >> the proxy. >> >> >> Thanks, >> Bill >> >> Brian West wrote: >> >>> it needs to be an ACL from acl.conf or a ip/cidr >>> >>> /b >>> >>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>> >>> >>>> Okay, I added: to my sofia >>>> profile and restarted sofia, and still no joy. >>>> >>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>> I've got in >>>> the directory, but I'm still being rejected by the acl: >>>> >>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>> Rejected by user acl 190.218.103.12/32 >>>> >>>> Here's what I believe is the appropriate snippet of the debug output: >>>> http://pastebin.freeswitch.org/11531 >>>> >>>> Thoughts? >>>> Thanks, >>>> Bill >>>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users-list at metik.com Thu Dec 17 20:21:22 2009 From: freeswitch-users-list at metik.com (Metik) Date: Thu, 17 Dec 2009 23:21:22 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AFEE8.5020002@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> Message-ID: <4B2B0342.3000201@metik.com> Why not simply implement this feature in the PROXY itself? FS has a pretty comprehensive security feature set for endpoints that directly register with it. Don't get me wrong, I do agree this is useful especially if you are going to be using your proxies to load balance across multiple FS boxes to create an ad-hoc cluster. I actually have session border controllers that have this feature and use it quite often. -metik Bill W wrote: > Hey Metik, > > Thanks for the reply, and the pointers for doing it with xml_curl. > > I'll guess have to do that in the short term, but in my opinion, having > auth-acl be able to work through a proxy is very important as it is a > vital part of a comprehensive security feature set. And it would be > much simpler to implement from an end-user perspective than the > alternative of doing it in xml_curl. > > As a matter of fact, I'm considering offering a bounty for that feature. > What is the going rate for that kind of thing? > > Is anyone out there interested in coding this feature? Or chipping in > for the bounty? > > > Thanks, > Bill > > > Metik wrote: > >> This may be difficult considering that ACL needs to consider the >> original src IP/URI. To do that it, freeswitch would need to do so >> using a header that retains that information (i.e. From, Via, Contact, >> etc.). Which I do not believe is currently possible using auth-acl or >> apply-proxy-acl. >> >> However, you should be able to emulate the behavior using mod_xml_curl >> (and validating against appropriate variables available when using it to >> authenticate the request). >> >> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >> >> -metik >> >> >> Bill W wrote: >> >>> Hey Brian, >>> >>> >>> I've been doing some testing and I am unable to get auth-calls to work >>> through a proxy the way I want them to, even with setting >>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>> >>> I have a multi-tenant system with multiple domains with multiple users >>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>> challenge them for a password. The arbitrary CIDR will vary from UA to >>> UA, and is specified in the directory via the auth-acl parameter. >>> >>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>> the proxy. >>> >>> >>> Thanks, >>> Bill >>> >>> Brian West wrote: >>> >>> >>>> it needs to be an ACL from acl.conf or a ip/cidr >>>> >>>> /b >>>> >>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>> >>>> >>>> >>>>> Okay, I added: to my sofia >>>>> profile and restarted sofia, and still no joy. >>>>> >>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>> I've got in >>>>> the directory, but I'm still being rejected by the acl: >>>>> >>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>> Rejected by user acl 190.218.103.12/32 >>>>> >>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>> http://pastebin.freeswitch.org/11531 >>>>> >>>>> Thoughts? >>>>> Thanks, >>>>> Bill >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mrene_lists at avgs.ca Thu Dec 17 22:08:16 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 18 Dec 2009 01:08:16 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2AFEE8.5020002@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> Message-ID: <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> From looking at sofia.c, if the ip address of the caller is in apply- proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, and use that one for authentication. Is that what you did in your previous tests? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 17-Dec-09, at 11:02 PM, Bill W wrote: > Hey Metik, > > Thanks for the reply, and the pointers for doing it with xml_curl. > > I'll guess have to do that in the short term, but in my opinion, > having > auth-acl be able to work through a proxy is very important as it is a > vital part of a comprehensive security feature set. And it would be > much simpler to implement from an end-user perspective than the > alternative of doing it in xml_curl. > > As a matter of fact, I'm considering offering a bounty for that > feature. > What is the going rate for that kind of thing? > > Is anyone out there interested in coding this feature? Or chipping in > for the bounty? > > > Thanks, > Bill > > > Metik wrote: >> This may be difficult considering that ACL needs to consider the >> original src IP/URI. To do that it, freeswitch would need to do so >> using a header that retains that information (i.e. From, Via, >> Contact, >> etc.). Which I do not believe is currently possible using auth-acl or >> apply-proxy-acl. >> >> However, you should be able to emulate the behavior using >> mod_xml_curl >> (and validating against appropriate variables available when using >> it to >> authenticate the request). >> >> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >> >> -metik >> >> >> Bill W wrote: >>> Hey Brian, >>> >>> >>> I've been doing some testing and I am unable to get auth-calls to >>> work >>> through a proxy the way I want them to, even with setting >>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>> >>> I have a multi-tenant system with multiple domains with multiple >>> users >>> in each domain. And I want to restrict a user to an arbitrary >>> CIDR and >>> challenge them for a password. The arbitrary CIDR will vary from >>> UA to >>> UA, and is specified in the directory via the auth-acl parameter. >>> >>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>> not of >>> the proxy. >>> >>> >>> Thanks, >>> Bill >>> >>> Brian West wrote: >>> >>>> it needs to be an ACL from acl.conf or a ip/cidr >>>> >>>> /b >>>> >>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>> >>>> >>>>> Okay, I added: to >>>>> my sofia >>>>> profile and restarted sofia, and still no joy. >>>>> >>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>> I've got >>>> param> in >>>>> the directory, but I'm still being rejected by the acl: >>>>> >>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>> 64.135.119.105 >>>>> Rejected by user acl 190.218.103.12/32 >>>>> >>>>> Here's what I believe is the appropriate snippet of the debug >>>>> output: >>>>> http://pastebin.freeswitch.org/11531 >>>>> >>>>> Thoughts? >>>>> Thanks, >>>>> Bill >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Thu Dec 17 22:30:32 2009 From: freeswitch at aastral.net (Bill W) Date: Fri, 18 Dec 2009 01:30:32 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2B0342.3000201@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> Message-ID: <4B2B2188.2060803@aastral.net> Hey Metik, That's exactly what I'm trying to do... load balance across multiple FS boxes, and have any machine in the cluster be able to reach a device behind a NAT firewall. Hence the need for the proxy. Also, I'm trying to keep the proxy relatively "dumb" and put all the logic in the FS boxes. True I could do the auth on the proxies as well, but then I'm setting up another authentication scheme in addition to what is on the FS boxes, and then integrating the databases so everything is consistent. I also have hosts that talk to the FS boxes directly, rather than through the proxy. So I can't get rid of auth_acl on FS either, even if I do implement it on the proxies. So my setup becomes much more complex and potentially brittle. And all we're really talking about for FreeSWITCH, conceptually speaking, is populating a variable with a different IP. We could even make it configurable, as to which IP is to be used for the auth-acl. What are you using for SBCs? (if you are allowed to divulge that) I'm currently using OpenSIPS for my proxy. Thanks, Bill Metik wrote: > Why not simply implement this feature in the PROXY itself? > > FS has a pretty comprehensive security feature set for endpoints that > directly register with it. > > Don't get me wrong, I do agree this is useful especially if you are > going to be using your proxies to load balance across multiple FS boxes > to create an ad-hoc cluster. I actually have session border controllers > that have this feature and use it quite often. > > -metik > > Bill W wrote: >> Hey Metik, >> >> Thanks for the reply, and the pointers for doing it with xml_curl. >> >> I'll guess have to do that in the short term, but in my opinion, having >> auth-acl be able to work through a proxy is very important as it is a >> vital part of a comprehensive security feature set. And it would be >> much simpler to implement from an end-user perspective than the >> alternative of doing it in xml_curl. >> >> As a matter of fact, I'm considering offering a bounty for that feature. >> What is the going rate for that kind of thing? >> >> Is anyone out there interested in coding this feature? Or chipping in >> for the bounty? >> >> >> Thanks, >> Bill >> >> >> Metik wrote: >> >>> This may be difficult considering that ACL needs to consider the >>> original src IP/URI. To do that it, freeswitch would need to do so >>> using a header that retains that information (i.e. From, Via, Contact, >>> etc.). Which I do not believe is currently possible using auth-acl or >>> apply-proxy-acl. >>> >>> However, you should be able to emulate the behavior using mod_xml_curl >>> (and validating against appropriate variables available when using it to >>> authenticate the request). >>> >>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>> >>> -metik >>> >>> >>> Bill W wrote: >>> >>>> Hey Brian, >>>> >>>> >>>> I've been doing some testing and I am unable to get auth-calls to work >>>> through a proxy the way I want them to, even with setting >>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>> >>>> I have a multi-tenant system with multiple domains with multiple users >>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>> UA, and is specified in the directory via the auth-acl parameter. >>>> >>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>> the proxy. >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> Brian West wrote: >>>> >>>> >>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>> >>>>> /b >>>>> >>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>> >>>>> >>>>> >>>>>> Okay, I added: to my sofia >>>>>> profile and restarted sofia, and still no joy. >>>>>> >>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>> I've got in >>>>>> the directory, but I'm still being rejected by the acl: >>>>>> >>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>> Rejected by user acl 190.218.103.12/32 >>>>>> >>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>> http://pastebin.freeswitch.org/11531 >>>>>> >>>>>> Thoughts? >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From darklion11 at yahoo.com Thu Dec 17 23:19:26 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 23:19:26 -0800 (PST) Subject: [Freeswitch-users] Destination Formats Expression Message-ID: <26840010.post@talk.nabble.com> Hi Everyone, Is there a link or tutorial for the expressions format. Example: 10 - default number [01[ - second number that start only on 0 or 1; [0-9] - 0 to 9 can be use Is there any? Thanks, Edmar -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840010.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Thu Dec 17 23:34:11 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 18 Dec 2009 18:34:11 +1100 Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <26840010.post@talk.nabble.com> References: <26840010.post@talk.nabble.com> Message-ID: <20091218073411.GA32288@jdc.jasonjgw.net> Edmar Cruz wrote: > > Is there a link or tutorial for the expressions format. Anything that describes Perl regular expressions should help, and for reference, see the pcre(3) manual page, and use the pcretest program to experiment. From lei.tlfly at gmail.com Thu Dec 17 23:42:46 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 18 Dec 2009 15:42:46 +0800 Subject: [Freeswitch-users] how does FS failover or load balance outbound calls between tow proxy Message-ID: <50c41b4e0912172342y67c4c8b2h1a99e41407b7eaaf@mail.gmail.com> Hi All I have a FS cluster behind two OpenSIPS proxy, the incoming calls is load balance and failover to FS cluster by OpenSips, It works well. The problem is, the outbound calls from FS must also route throw then OpenSIPS servers. So, does FS servers can loadbalance the outbound calls between the two OpenSIPS servers and failover if one of the Opensips server is down? -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/443ea268/attachment-0002.html From msc at freeswitch.org Thu Dec 17 23:51:26 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 17 Dec 2009 23:51:26 -0800 Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <20091218073411.GA32288@jdc.jasonjgw.net> References: <26840010.post@talk.nabble.com> <20091218073411.GA32288@jdc.jasonjgw.net> Message-ID: <840F0EA0-227F-4CCD-BCB8-2F4945205A70@freeswitch.org> On Dec 17, 2009, at 11:34 PM, Jason White wrote: > Edmar Cruz wrote: >> >> Is there a link or tutorial for the expressions format. > > Anything that describes Perl regular expressions should help, and for > reference, see the pcre(3) manual page, and use the pcretest program > to > experiment. > http://wiki.freeswitch.org/wiki/Regular_Expression -MC From darklion11 at yahoo.com Thu Dec 17 23:58:17 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 17 Dec 2009 23:58:17 -0800 (PST) Subject: [Freeswitch-users] Destination Formats Expression In-Reply-To: <20091218073411.GA32288@jdc.jasonjgw.net> References: <26840010.post@talk.nabble.com> <20091218073411.GA32288@jdc.jasonjgw.net> Message-ID: <26840254.post@talk.nabble.com> Thanks that will be a great help Jason White-14 wrote: > > Edmar Cruz wrote: >> >> Is there a link or tutorial for the expressions format. > > Anything that describes Perl regular expressions should help, and for > reference, see the pcre(3) manual page, and use the pcretest program to > experiment. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Destination-Formats-Expression-tp26840010p26840254.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From djbinter at yahoo.com Fri Dec 18 00:10:29 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 00:10:29 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> Message-ID: <922386.16417.qm@web37502.mail.mud.yahoo.com> Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, > >I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >Please advise if you need further info. > >Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/7ac70921/attachment-0002.html From devel at thom.fr.eu.org Fri Dec 18 01:51:08 2009 From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=) Date: Fri, 18 Dec 2009 10:51:08 +0100 Subject: [Freeswitch-users] Voicemail->Email Message-ID: I get the same result with sendmail. This used to work in 1.0.3 , and after upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is still there. Fran?ois On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Sch?nbeck wrote: Currently it is Version 1.0.trunk (15982) VON: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] IM AUFTRAG VON Brian West GESENDET: Donnerstag, 17. Dezember 2009 17:17 AN: freeswitch-users at lists.freeswitch.org BETREFF: Re: [Freeswitch-users] Voicemail->Email What SVN rev. exactly? /b On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: Hello, we are running freeswitch 1.0.trunk and are currently trying to get the mod_voicemail to send the received messages to the user by using exim4 on a debian machine. So far we followed the instructions in the wiki article ( http://wiki.freeswitch.org/wiki/Mod_voicemail ). I added some lines to the bash script to enable some kind of logging: #! /bin/bash typeset LOG="/tmp/${0##*/}.out" mv $LOG ${LOG}.old >/dev/null 2>"> [[ -t 1 ]] &">exec > $LOG 2>">exim4 -t -v >> $LOG If I run the script from the command line everything is working as expected. If the script gets called by freeswitch I get the following result in my logfile: /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation fault (core dumped) exim4 -t -v >> $LOG Has anybody seen similar effects before? Any advice whats going wrong is heavily appreciated. Thanks Oliver _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org [1] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [2] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [3] Links: ------ [1] mailto:FreeSWITCH-users at lists.freeswitch.org [2] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [3] http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/3292d179/attachment-0002.html From Prometheus001 at gmx.net Fri Dec 18 02:00:55 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 18 Dec 2009 11:00:55 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: <4B28D6FD.6010702@gmx.net> References: <4B28D6FD.6010702@gmx.net> Message-ID: <4B2B52D7.9030505@gmx.net> Should I open a JIRA for this? Best regards Peter Peter P GMX schrieb: > Hello, > > we have the following scenario: > A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For > the called FS user, call forwarding has been enabled to another PSTN > extension (B) . > Result: The calling party does not hear any ringing tone. Here an > Abstract of the SIP protocol we traced (PSTN(A) and PSTN(B) is in fact > the same Patton Gateway): > > PSTN(A)====INVITE===>FS > PSTN(A)<===TRYING===>FS > FS===INVITE==>PSTN(B) > FS<==TRYING===PSTN(B) > FS<==RINGING==PSTN(B) > PSTN(A)<==PROGRESS===FS > FS<===OK======PSTN(B) > FS====ACK====>PSTN(B) > PSTN(A)<===OK========FS > PSTN(A)====ACK======>FS > > I would expect that FS answers RINGING back to PSTN(A). Instead it only > answers SESSION PROGRESS. > When PSTN(B) answers, they can hear each other, but there was no ringing > tone to PSTN(A) before. > > Are there any hints to overcome this, besides playing early media to > PSTN(A)? > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From viper at fx-services.com Fri Dec 18 02:21:12 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 18 Dec 2009 11:21:12 +0100 Subject: [Freeswitch-users] LUA and return variables Message-ID: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> Hi guys (and girls)! I'm working on a little bit of ENUM trickery and I tried doing some (illegal) nested conditions. :-) What I want to do is to first check enum with the ENUM application, then depending on the answer do some stuff. Say that the domain part of the ENUM answer is robin.nl, then I want to do action X instead of just briding the enum answer directly as I see in most examples. But I remembered that it wasn't allowed to do nested conditions. So what I did was stacked conditions. After that I read the dialplan wiki pages again and figured that my regexp never matches because variables I "set" during some phase of the extension I can't use in the same "go" as another condition. So, now my plan is to use LUA to do the regexp. I'll pass the enum answer to a lua script which will split the answer in a user and domain part and return those two to the main app. Then based on those two vars I'll do routing or other actions (like, prefix and then route). Is this how I'm supposed to do it? I can't find many examples on manipulating ENUM answers, other than bridging them directly. I can't change the way I do stuff to ENUM answers, because in most cases I'll just route them out the standard way. Anyone with experience on fiddling with ENUM answers? -- Robin Vleij From bcxml at hotmail.com Fri Dec 18 03:16:12 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 18 Dec 2009 03:16:12 -0800 (PST) Subject: [Freeswitch-users] Ringing after call has been rejected Message-ID: <26842055.post@talk.nabble.com> I have an incomming call being answered by FreeSwitch and passed to IVR application which rejects the call. The call is never answered by FreeSwitch, but instead of hearing a busy signal, the caller hears ringing. Can anyone advise how I can get the user to hear a busy signal after call rejection instead of ringing. Here is the debug trace http://pastebin.freeswitch.org/11558 Thanks Brian -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Fri Dec 18 05:04:51 2009 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Dec 2009 13:04:51 +0000 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <191c3a030912171446j45505417u5db6218243d0bc4c@mail.gmail.com> Message-ID: Brian, You haven't said what codecs are being used yet. Are the listeners using a different codec to the speaker? If so, you're potentially doing transcoding on every single channel, which would make CPU usage skyrocket. -Steve 2009/12/17 Anthony Minessale : > What exactly is your test process? > > you should try increasing the interval in the conference profile to a bigger > time slice maybe 30 40 or 60ms > you could also increase the ptime to match as well. > > > like brian said you could use mod_shout to broadcast the single speaker to > icecast and let people listen with itunes/winamp > > > On Thu, Dec 17, 2009 at 3:41 PM, Brian wrote: >> >> I did a test with the trunk version for the one conference case, and it is >> the same results as for 1.0.4. The audio failed at around 300 listeners. >> Oddly though, it consumed less %CPU (240% instead of 300%), and yet the >> audio still failed at the same number of listeners. >> >> >> >> Brian. >> >> >> >> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] >> Sent: Thursday, December 17, 2009 3:49 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> We didn't post it anywhere but we just get overwhelmed with them and many >> of them are unfounded and take up a lot of time to track down.? That does >> not mean you have not found a real problem but the first step is trying >> trunk. >> >> >> On Thu, Dec 17, 2009 at 2:32 PM, Brian wrote: >> >> I didn?t realize there was a policy about load testing questions. What >> forum should I have used for this? >> >> >> >> I didn?t get the chance to test on FS trunk yet, but when I do I will >> provide you with the feedback when I do. Just let me know what forum to use >> for this topic from now on. >> >> >> >> Thanks, >> >> >> >> Brian. >> >> >> >> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] >> Sent: Thursday, December 17, 2009 2:42 PM >> >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> One man's stable release is another man's 6 month old release with >> hundreds of known fixed bugs. >> If one of the core developers tells you to try it, you may as well take >> the time to try it now that you have opened a forum questioning the >> scalability. >> >> When you tested asterisk did you actually use 600 phones and verify that >> each one can hear the audio perfectly and in time with what the speaker was >> saying?? Did you try same on FS? >> >> Did you optimize your dialplan on FS to deal with a load test or follow >> any of the recommended performance tuning page. >> >> All of the answers to these questions are really moot because we have a >> policy against entertaining load testing questions but if you like asterisk, >> by all means, use it, and good luck to you if those numbers you are testing >> at are what you plan to put in real production......... >> >> On Thu, Dec 17, 2009 at 1:29 PM, Brian wrote: >> >> Hi Mike, >> >> >> >> I didn?t get around to testing on the FreeSWITCH trunk yet. Are there >> substantial fixes to mod_conference in the FreeSWITCH trunk that might >> increase capacity for my scenario of one speaker and many listeners? If I >> want to put this into a production environment, I would need a stable >> version, which as far as I know is the 1.0.4 version. >> >> >> >> However, I did test on Asterisk 1.4 using app_conference, and doing the >> same scenario was able to get 1 speaker and 600 listeners on a single >> conference with no audio issues. The CPU at that point was just over 300%, >> same as where the single conference scenario failed on FreeSWITCH with 300 >> listeners. ?I was able to push it to over 700 listeners before I reached >> 400% CPU usage (I guess maxing out my quad-core processors), and asterisk >> finally crashed. But up until that point, there were no audio problems. >> >> >> >> I?ve read a lot about how FreeSWITCH is supposed to be more scalable than >> Asterisk, but unless there is something wrong with my FreeSWITCH setup, >> Asterisk was clearly the winner in this test ? more than doubling FreeSWITCH >> capacity in this case. Again, maybe there is something on the FreeSWITCH >> side that I?m doing wrong, but I don?t see what it could be. >> >> >> >> Brian. >> >> >> >> >> >> From: Michael Jerris [mailto:mike at jerris.com] >> Sent: Thursday, December 17, 2009 10:18 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] mod_conference scalability >> >> >> >> I would be curious what the same tests produce with svn trunk of >> FreeSWITCH. >> >> >> >> Mike >> >> >> >> On Dec 16, 2009, at 4:49 PM, Brian wrote: >> >> >> >> Hi, >> >> >> >> I?m new to FreeSWITCH and I?m testing the scalability of mod_conference to >> see if it will scale better that other solutions. My scenario is to have one >> speaker, and many listeners (mute). Since I have only one speaker, I was >> expecting this to scale well because there is no audio mixing required, just >> send each frame of the single speaker to each listener. Unfortunately, my >> testing was disappointing, and it didn?t scale nearly as well as I?d hoped >> (based on what I?ve read on how FreeSWITCH is supposed to be generally very >> scalable). >> >> >> >> Here?s my server setup is this: >> >> >> >> FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of >> RAM. I?ve set file logging to ?notice? level. My conference profile is >> configured to suppress several events, hoping that it would improve >> performance. >> >> >> >> Here are a few scenarios I tested, and roughly where I reached the point >> of audio failure on the conferences: >> >> >> >> Scenario 1: >> >> 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) >> >> >> >> Scenario 2: >> >> 4 conferences, 1 speaker per conference, audio failed approx 110 listeners >> per conference (so just over 400 total channels on the system). >> >> >> >> Scenario 3: >> >> 16 conferences, 1 speaker per conference, audio failed at 32 listeners per >> conference (so just over 500 total channels on the system). >> >> >> >> >> >> Looking at the output from ?top?, it seems that in all 3 scenarios, the >> audio quality failed when the % CPU for the FreeSWITCH process exceeded >> 300%. >> >> >> >> I was hoping maybe someone else might have done similar testing, or maybe >> has suggestions on how to improve the performance. Or perhaps an alternate >> solution to the one speaker, many listener case? >> >> >> >> Thanks, >> >> >> >> Brian. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From keith at laaks.com Fri Dec 18 05:13:48 2009 From: keith at laaks.com (Keith Laaks) Date: Fri, 18 Dec 2009 15:13:48 +0200 (SAST) Subject: [Freeswitch-users] mod_xml_ldap compile issue. Message-ID: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> Hi, I am having an issue getting mod_xml_ldap to compile properly.... making all mod_xml_cdr making all mod_xml_ldap Creating mod_xml_ldap.la... /usr/bin/ld: /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: could not read symbols: Bad value collect2: ld returned 1 exit status cat: .libs/mod_xml_ldap.log: No such file or directory make[5]: *** [mod_xml_ldap.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_ldap-all] Error 1 make[2]: *** [all-recursive] Error 1 ------------------------------------------------------------------------ I notice the openldap library has been bumped up to .19 - not sure if that may have anything to do with it. At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. mod_ldap compiles OK, but mod_xml_ldap fails as per the above. What am I doing working here ? Best Regards Keith From brian at proximosystems.com Fri Dec 18 06:08:31 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 09:08:31 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> Message-ID: <007101ca7feb$9462de70$bd289b50$@com> I've got FS running on a 64 bit OS, and here is more info on the test procedure. I've got one server (primary) that hosts the speaker call (this is meant to be a primary conference with a few speakers, but my test simplifies this to just one speaker). I've got a second server (secondary) that hosts the conference that all the listeners go into, and I have two other servers that I use automate the listener calls. The goal is to have several secondary servers to scale the listener side of things, but for this initial test I've only got one secondary server. The primary server dials into the secondary conference server so that the listeners can hear the speaker conference on the primary server. The automated listener servers start dialing into the listener conference at a combined rate of 5 calls per second (i.e. 2.5 calls per second each). The play an audio loop that represents noise on their end, which since they are listeners, should be ignored anyway. As I ramp up the automated listener calls, I manually call into the conference from either my SIP phone, or from a land line using a DID that I have directed to the conference. All calls are using SIP with uLaw 8000hz codec. Also, I've set up the profile for the listener conference to disable many of the events: I do have caller controls for the listener, since in my production I will need to generate and handle events for listener DTMF. To compare FreeSWITCH vs Asterisk, I just swap out the secondary conference server and everything else stays the same. Brian. From: Brian West [mailto:brian at freeswitch.org] Sent: Thursday, December 17, 2009 5:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability What exactly are you doing I know it goes better than that.. are you using 64bit? / b On Dec 17, 2009, at 3:41 PM, Brian wrote: I did a test with the trunk version for the one conference case, and it is the same results as for 1.0.4. The audio failed at around 300 listeners. Oddly though, it consumed less %CPU (240% instead of 300%), and yet the audio still failed at the same number of listeners. Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/8845d8d1/attachment-0002.html From brian at freeswitch.org Fri Dec 18 06:43:27 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Dec 2009 08:43:27 -0600 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: <4B2B52D7.9030505@gmx.net> References: <4B28D6FD.6010702@gmx.net> <4B2B52D7.9030505@gmx.net> Message-ID: That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. /b On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: > Should I open a JIRA for this? > > Best regards > Peter From yhding2003 at yahoo.ca Fri Dec 18 06:54:32 2009 From: yhding2003 at yahoo.ca (yvonne ding) Date: Fri, 18 Dec 2009 06:54:32 -0800 (PST) Subject: [Freeswitch-users] How can I join two freeswitch on two servers? In-Reply-To: <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> References: <24045824.post@talk.nabble.com> <914fc92a0906161443h417047a4sa92dd195d43705de@mail.gmail.com> <26831042.post@talk.nabble.com> <26832823.post@talk.nabble.com> <23f91030912171736m1fc44a5dq90e2c776d4711325@mail.gmail.com> Message-ID: <26844589.post@talk.nabble.com> Hi, As far as I know, there are two ways to connect two freeswitch, by using ACL or using authentication. Also from this email history discussion, another solution is to create user in FS B directory,then treat server B as normal gateway by adding gateway definiton in FS A. So my question is how to connect FS A and FS B through ACL or through the way this email described. The information I pasted is about the last way. FS A: 192.168.129.168, caller id= 1001 FS B: 192.168.129.194, callee id= 1003, create 1101 for gateway configure In FS A add /conf/sip_proifles/external/gwfsa.xml param name="username" value="1101" param name="password" value="1234" param name="proxy" value="192.168.129.194:5060" param name="register" value="false" note: I delete < and /> for param cause it can't be displayed in this email. Both FS A and FS B are default configuration except creating id=1101 on FS B side. I'm confused if I connect two freeswitch by using ACLs, How do I confiugre data in both side ? Your kind help is highly appreciated. Seven Du wrote: > > I couldn't guess what you want, pastbin your full config and logs and > give more detail of your story perhaps someone can help you. > > 2009/12/18 yvonne ding : >> >> param name="username" value="1101" >> param name="password" value="1234" >> param name="proxy" value="192.168.129.194:5060" >> param name="register" value="false" >> >> >> Hi, >> >> If I configure data as following, why FS A "1001" call FS B "1003" failed >> ? >> Thank you! >> >> FS A: 192.168.129.168, DN=1001 >> FS B: 192.168.129.194, DN=1003 >> >> In FS A add /conf/sip_proifles/external/gwfsa.xml >> ? >> ? ? >> >> >> >> >> ? ? >> >> >> 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't >> have >> 1101 number >> >> >> >> >> >> Dan Le wrote: >>> >>> If you want FS server A to be able to call FS server B, you can set up a >>> user account in server B's FS directory configs, and then just treat >>> server >>> B as a normal gateway by adding a gateway definition in server A. That >>> will >>> allow you to route calls to server B from A; to do the reverse, just >>> mirror >>> the configs the other direction. >>> >>> On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz >>> wrote: >>> >>>> >>>> I like to connect two freeswitch, call each other, communicate and vice >>>> versa. >>>> Can you give me an example for that? >>>> >>>> Thanks >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> View this message in context: >> http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26832823.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p26844589.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jbr at consiglia.dk Fri Dec 18 07:23:27 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 18 Dec 2009 16:23:27 +0100 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: I have found some ways to get presence, or rather BLF functions to work on Snom telephones in a distributed network with several FSs. I'll post a solution on the wiki when I have tested it further. Anyhow, I'm using the mod_event_multicast module with the following configuration: With this setting on all FSs, the registration table is also automatically updated thus listing all sets registered across all FSs. In the table sip_registrations (under the database for the profile used), the field status has the value: "Registered" if the UA is registered on another FS and the value "Registered(UDP)" if the UA is registered on the same FS. The field server_host, however, is the ip-address of "local" FS. Now comes the question: is there any way to let the field server_host show the server address of the server actually registered to? Or any other way using the existing modules to get the information about which FS the UAs are registered to? The information is going to be used for the routing decisions between networked FSs. /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/fbe8a447/attachment-0002.html From freeswitch-list at puzzled.xs4all.nl Fri Dec 18 07:29:35 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Fri, 18 Dec 2009 16:29:35 +0100 Subject: [Freeswitch-users] mod_xml_ldap compile issue. In-Reply-To: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> References: <47263.196.41.30.5.1261142028.squirrel@mail.laaks.com> Message-ID: <4B2B9FDF.9060600@puzzled.xs4all.nl> On 12/18/2009 02:13 PM, Keith Laaks wrote: > Hi, > > I am having an issue getting mod_xml_ldap to compile properly.... > > > making all mod_xml_cdr > > making all mod_xml_ldap > Creating mod_xml_ldap.la... > /usr/bin/ld: > /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a(sasl.o): > relocation R_X86_64_32S against `.rodata' can not be used when making a > shared object; recompile with -fPIC > /home/keithl/freeswitch/freeswitch.trunk/libs/openldap-2.4.19/libraries/liblutil/liblutil.a: > could not read symbols: Bad value > collect2: ld returned 1 exit status > cat: .libs/mod_xml_ldap.log: No such file or directory > make[5]: *** [mod_xml_ldap.la] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_xml_ldap-all] Error 1 > make[2]: *** [all-recursive] Error 1 > ------------------------------------------------------------------------ > I notice the openldap library has been bumped up to .19 - not sure if that > may have anything to do with it. > > At revision 15995 on a 2.6.31-15-generic Ubuntu x86_64 GNU/Linux notebook. > > mod_ldap compiles OK, but mod_xml_ldap fails as per the above. > > What am I doing working here ? I had the same issue and MikeJ (one of the core developers) looked at it. Conclusion was that it is an openldap issue and iirc the solution is to libtoolize libraries/liblutil/Makefile.in so that when running configure a Makefile with proper compiler flags is generated in libraries/liblutil/ Patches welcome :) Regards, Patrick From mike at jerris.com Fri Dec 18 07:29:13 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Dec 2009 10:29:13 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <007101ca7feb$9462de70$bd289b50$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <99F9804D-D855-4419-8880-51276A1B4FE6@freeswitch.org> <007101ca7feb$9462de70$bd289b50$@com> Message-ID: What is your dialplan on the secondary box? On Dec 18, 2009, at 9:08 AM, Brian wrote: > I?ve got FS running on a 64 bit OS, and here is more info on the tes > t procedure. > > > > I?ve got one server (primary) that hosts the speaker call (this is m > eant to be a primary conference with a few speakers, but my test sim > plifies this to just one speaker). I?ve got a second server (seconda > ry) that hosts the conference that all the listeners go into, and I > have two other servers that I use automate the listener calls. The g > oal is to have several secondary servers to scale the listener side > of things, but for this initial test I?ve only got one secondary ser > ver. > > > > The primary server dials into the secondary conference server so > that the listeners can hear the speaker conference on the primary > server. > > > > The automated listener servers start dialing into the listener > conference at a combined rate of 5 calls per second (i.e. 2.5 calls > per second each). The play an audio loop that represents noise on > their end, which since they are listeners, should be ignored anyway. > > > > As I ramp up the automated listener calls, I manually call into the > conference from either my SIP phone, or from a land line using a DID > that I have directed to the conference. > > > > All calls are using SIP with uLaw 8000hz codec. Also, I?ve set up th > e profile for the listener conference to disable many of the events: > > > > > > > > > > > > > > > > > > > > I do have caller controls for the listener, since in my production I > will need to generate and handle events for listener DTMF. > > > > To compare FreeSWITCH vs Asterisk, I just swap out the secondary > conference server and everything else stays the same. > > > > Brian. > > > > From: Brian West [mailto:brian at freeswitch.org] > Sent: Thursday, December 17, 2009 5:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > What exactly are you doing I know it goes better than that.. are you > using 64bit? > > > > / b > > > > On Dec 17, 2009, at 3:41 PM, Brian wrote: > > > > > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c45449a5/attachment-0002.html From freeswitch at aastral.net Fri Dec 18 07:53:11 2009 From: freeswitch at aastral.net (Bill W) Date: Fri, 18 Dec 2009 10:53:11 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> Message-ID: <4B2BA567.6010202@aastral.net> Hello Mathieu, I assumed that apply-proxy-acl was a modifier of auth-calls, so in my quick tests I just hard-coded the UA IP in the profile. And I get: 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 Rejected by user acl 190.218.97.83/32 Where 64.135.119.105 is the IP of my proxy. And actually this is a REGISTER, not an INVITE. I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the register packet. I will be incommunicado for the rest of today, but when I get back online, I'll see if I can get my proxy to add the X-AUTH-IP to the REGISTER packet and see if that makes a difference. Thanks for your help! Bill Mathieu Rene wrote: > From looking at sofia.c, if the ip address of the caller is in apply- > proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, > and use that one for authentication. > Is that what you did in your previous tests? > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 17-Dec-09, at 11:02 PM, Bill W wrote: > >> Hey Metik, >> >> Thanks for the reply, and the pointers for doing it with xml_curl. >> >> I'll guess have to do that in the short term, but in my opinion, >> having >> auth-acl be able to work through a proxy is very important as it is a >> vital part of a comprehensive security feature set. And it would be >> much simpler to implement from an end-user perspective than the >> alternative of doing it in xml_curl. >> >> As a matter of fact, I'm considering offering a bounty for that >> feature. >> What is the going rate for that kind of thing? >> >> Is anyone out there interested in coding this feature? Or chipping in >> for the bounty? >> >> >> Thanks, >> Bill >> >> >> Metik wrote: >>> This may be difficult considering that ACL needs to consider the >>> original src IP/URI. To do that it, freeswitch would need to do so >>> using a header that retains that information (i.e. From, Via, >>> Contact, >>> etc.). Which I do not believe is currently possible using auth-acl or >>> apply-proxy-acl. >>> >>> However, you should be able to emulate the behavior using >>> mod_xml_curl >>> (and validating against appropriate variables available when using >>> it to >>> authenticate the request). >>> >>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>> >>> -metik >>> >>> >>> Bill W wrote: >>>> Hey Brian, >>>> >>>> >>>> I've been doing some testing and I am unable to get auth-calls to >>>> work >>>> through a proxy the way I want them to, even with setting >>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>> >>>> I have a multi-tenant system with multiple domains with multiple >>>> users >>>> in each domain. And I want to restrict a user to an arbitrary >>>> CIDR and >>>> challenge them for a password. The arbitrary CIDR will vary from >>>> UA to >>>> UA, and is specified in the directory via the auth-acl parameter. >>>> >>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>>> not of >>>> the proxy. >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> Brian West wrote: >>>> >>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>> >>>>> /b >>>>> >>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>> >>>>> >>>>>> Okay, I added: to >>>>>> my sofia >>>>>> profile and restarted sofia, and still no joy. >>>>>> >>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>> I've got >>>>> param> in >>>>>> the directory, but I'm still being rejected by the acl: >>>>>> >>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>>> 64.135.119.105 >>>>>> Rejected by user acl 190.218.103.12/32 >>>>>> >>>>>> Here's what I believe is the appropriate snippet of the debug >>>>>> output: >>>>>> http://pastebin.freeswitch.org/11531 >>>>>> >>>>>> Thoughts? >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mrene_lists at avgs.ca Fri Dec 18 07:55:52 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 18 Dec 2009 10:55:52 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2BA567.6010202@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <00D5CDC4-3753-4192-9937-A2966EAF7EA8@avgs.ca> <4B2BA567.6010202@aastral.net> Message-ID: <11616732-C3B1-4B03-9368-2C777C402F1F@avgs.ca> You need to add that header manually in your OpenSIPS config, FreeSWITCH wont look in record-route/via to try to guess it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 18-Dec-09, at 10:53 AM, Bill W wrote: > Hello Mathieu, > > I assumed that apply-proxy-acl was a modifier of auth-calls, so in my > quick tests I just hard-coded the UA IP in the profile. > > > > > And I get: > 2009-12-18 09:14:28.250929 [WARNING] sofia_reg.c:1928 IP > 64.135.119.105 > Rejected by user acl 190.218.97.83/32 > > Where 64.135.119.105 is the IP of my proxy. And actually this is a > REGISTER, not an INVITE. > > I did a tcpdump, and I'm not seeing the X-AUTH-IP header in the > register > packet. > > I will be incommunicado for the rest of today, but when I get back > online, I'll see if I can get my proxy to add the X-AUTH-IP to the > REGISTER packet and see if that makes a difference. > > > Thanks for your help! > Bill > > > Mathieu Rene wrote: >> From looking at sofia.c, if the ip address of the caller is in apply- >> proxy-acl, it'll look for the X-AUTH-IP header in the INVITE packet, >> and use that one for authentication. >> Is that what you did in your previous tests? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 17-Dec-09, at 11:02 PM, Bill W wrote: >> >>> Hey Metik, >>> >>> Thanks for the reply, and the pointers for doing it with xml_curl. >>> >>> I'll guess have to do that in the short term, but in my opinion, >>> having >>> auth-acl be able to work through a proxy is very important as it >>> is a >>> vital part of a comprehensive security feature set. And it would be >>> much simpler to implement from an end-user perspective than the >>> alternative of doing it in xml_curl. >>> >>> As a matter of fact, I'm considering offering a bounty for that >>> feature. >>> What is the going rate for that kind of thing? >>> >>> Is anyone out there interested in coding this feature? Or chipping >>> in >>> for the bounty? >>> >>> >>> Thanks, >>> Bill >>> >>> >>> Metik wrote: >>>> This may be difficult considering that ACL needs to consider the >>>> original src IP/URI. To do that it, freeswitch would need to do so >>>> using a header that retains that information (i.e. From, Via, >>>> Contact, >>>> etc.). Which I do not believe is currently possible using auth- >>>> acl or >>>> apply-proxy-acl. >>>> >>>> However, you should be able to emulate the behavior using >>>> mod_xml_curl >>>> (and validating against appropriate variables available when using >>>> it to >>>> authenticate the request). >>>> >>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>> >>>> -metik >>>> >>>> >>>> Bill W wrote: >>>>> Hey Brian, >>>>> >>>>> >>>>> I've been doing some testing and I am unable to get auth-calls to >>>>> work >>>>> through a proxy the way I want them to, even with setting >>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>> >>>>> I have a multi-tenant system with multiple domains with multiple >>>>> users >>>>> in each domain. And I want to restrict a user to an arbitrary >>>>> CIDR and >>>>> challenge them for a password. The arbitrary CIDR will vary from >>>>> UA to >>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>> >>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, >>>>> not of >>>>> the proxy. >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> Brian West wrote: >>>>> >>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>> >>>>>> /b >>>>>> >>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>> >>>>>> >>>>>>> Okay, I added: to >>>>>>> my sofia >>>>>>> profile and restarted sofia, and still no joy. >>>>>>> >>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>> I've got >>>>>> param> in >>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>> >>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP >>>>>>> 64.135.119.105 >>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>> >>>>>>> Here's what I believe is the appropriate snippet of the debug >>>>>>> output: >>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>> >>>>>>> Thoughts? >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Dec 18 07:56:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 09:56:02 -0600 Subject: [Freeswitch-users] Voicemail->Email In-Reply-To: References: Message-ID: <191c3a030912180756i613e67d4ha5fef2142a70f82d@mail.gmail.com> oh really, sendmail segfaults? if another application is crashing you need to figure that out, whatever used to work doesnt now so you need to figure out what it was and let us know. On Fri, Dec 18, 2009 at 3:51 AM, Fran?ois Legal wrote: > I get the same result with sendmail. This used to work in 1.0.3 , and after > upgrading to 1.0.4 (then some snapshots) then 1.0.5pre8 and the problem is > still there. > > > > Fran?ois > > > > On Thu, 17 Dec 2009 17:33:58 +0100, Oliver Sch?nbeck wrote: > > Currently it is Version 1.0.trunk (15982) > > > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Brian West > *Gesendet:* Donnerstag, 17. Dezember 2009 17:17 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] Voicemail->Email > > > > What SVN rev. exactly? > > > > /b > > > > On Dec 17, 2009, at 10:13 AM, Oliver Sch?nbeck wrote: > > > > Hello, > > > > we are running freeswitch 1.0.trunk and are currently trying to get the > mod_voicemail to send the received messages to the user by using exim4 on a > debian machine. > > > > So far we followed the instructions in the wiki article ( > http://wiki.freeswitch.org/wiki/Mod_voicemail ). > > > > I added some lines to the bash script to enable some kind of logging: > #! /bin/bash > > typeset LOG="/tmp/${0##*/}.out" > > mv $LOG ${LOG}.old >/dev/null 2>&1 > > [[ -t 1 ]] && echo "Writing to logfile '$LOG'." > > exec > $LOG 2>&1 > > exim4 -t -v >> $LOG > > > > If I run the script from the command line everything is working as > expected. If the script gets called by freeswitch I get the following result > in my logfile: > > /usr/local/freeswitch/scripts/exec_exim.sh: line 6: 4920 Segmentation > fault (core dumped) exim4 -t -v >> $LOG > > > > Has anybody seen similar effects before? > > > > Any advice whats going wrong is heavily appreciated. > > > > Thanks > > Oliver > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/93835bd7/attachment-0002.html From fdelawarde at wirelessmundi.com Fri Dec 18 08:12:08 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 18 Dec 2009 17:12:08 +0100 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <04ca01ca7f61$a0560e30$e1022a90$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> Message-ID: <1261152728.11815.57.camel@luna.tc.commsmundi.com> Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) Fran?ois. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 3:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > We didn't post it anywhere but we just get overwhelmed with them and > many of them are unfounded and take up a lot of time to track down. > That does not mean you have not found a real problem but the first > step is trying trunk. > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Fri Dec 18 08:34:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 10:34:21 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261152728.11815.57.camel@luna.tc.commsmundi.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> Message-ID: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:888 at conference.freeswitch.orgThis is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > > substantial fixes to mod_conference in the FreeSWITCH trunk that might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > > than Asterisk, but unless there is something wrong with my FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other solutions. My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the single > > speaker to each listener. Unfortunately, my testing was disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > > of RAM. I?ve set file logging to ?notice? level. My conference profile > > is configured to suppress several events, hoping that it would improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/8ba43e55/attachment-0002.html From msc at freeswitch.org Fri Dec 18 08:54:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 08:54:23 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Starting Shortly! Message-ID: <87f2f3b90912180854h59b651c2t289f5a42ebec3973@mail.gmail.com> Hello everyone! Today's agenda is listed here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14 Also, we are going to be giving away goodies on some of the upcoming conferences, so call in and see what we've got in store. :) For the first 15 minutes we'll let everyone mingle and then we'll get into the agenda. Talk to you all soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/df3e27df/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 18 09:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 11:21:19 -0600 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <26842055.post@talk.nabble.com> References: <26842055.post@talk.nabble.com> Message-ID: <191c3a030912180921t4f188291w5f36ff3eb647f924@mail.gmail.com> not answering it would be the best way. if you want to generate fake congestion you can use tone_stream:// or gentones On Fri, Dec 18, 2009 at 5:16 AM, bcxml wrote: > > I have an incomming call being answered by FreeSwitch and passed to IVR > application which rejects the call. > > The call is never answered by FreeSwitch, but instead of hearing a busy > signal, the caller hears ringing. > > Can anyone advise how I can get the user to hear a busy signal after call > rejection instead of ringing. > > Here is the debug trace > > http://pastebin.freeswitch.org/11558 > > Thanks > > > Brian > > -- > View this message in context: > http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/60f3ddf7/attachment-0002.html From srinivas.ksvreddy at gmail.com Thu Dec 17 23:41:08 2009 From: srinivas.ksvreddy at gmail.com (srinivasula reddy) Date: Fri, 18 Dec 2009 13:11:08 +0530 Subject: [Freeswitch-users] Fwd: incoming call In-Reply-To: References: Message-ID: Hi, i have up the freeswitch with domain(eg sipserver.domain.com) name instead of local ip, two clints are regitered with freeswitch using domain name(eg sipserver.domain.com), one client is making a call to other one, other clint receiving a invite request like this 173927 3120.658532 10.91.154.108 10.91.154.80 SIP/SDP Request: INVITE sip:1010 at 10.91.154.80:5061, with session description, but usually it should come with INVITE sip:1010 at sipserver.domain.com? what is changes i need to do for this? any idea? Regards-- Srinivasula Reddy K -- Srinivasula Reddy K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/ca16ea09/attachment-0002.html From mike at jerris.com Fri Dec 18 09:37:16 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Dec 2009 12:37:16 -0500 Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <922386.16417.qm@web37502.mail.mud.yahoo.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> Message-ID: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: > Mike, > > My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > Thank you, > Dorn B. > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thu, December 17, 2009 8:03:46 AM > Subject: Re: [Freeswitch-users] SIP Re-invite > > are you doing this trace from the freeswitch box itself? > > Mike > > On Dec 17, 2009, at 10:48 AM, DJB wrote: > >> Anthony, >> >> I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >> Please advise if you need further info. >> >> Thank you. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/3a50f5f6/attachment-0002.html From fdelawarde at wirelessmundi.com Fri Dec 18 09:41:44 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Fri, 18 Dec 2009 18:41:44 +0100 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Message-ID: <1261158104.11815.92.camel@luna.tc.commsmundi.com> It was of course just bad humor, I love both projects for what they are, and I agree that both have their own advantages and inconvenients. For example, accessing that same conference from a dahdi card could be another goal where Asterisk would be at an advantage, as chan_dahdi is still superior (in the more tested sense) than openzap+mod_openzap. I just use both projects separately or together depending on what's needed! I'm no banker nor do I understand the code, but many thanks for all those unpaid contributions providing an excellent alternative for free telephony. Your names really deserve being engraved in google's cache for eternity. :-) But still, I would like to see those numbers... Fran?ois. On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to sponsor > the effort to add features to the conference module, but of course, I > do not hold my breath, instead I continue to improve it for free when > I find time. This is one of many reasons I do not enjoy performance > discussions unless I am talking to an engineer who understands the > code or a banker ready to pay for improvements. That is not my way of > saying pay me or forget it as you can clearly see the conference > module has made it to where it is today with no financial support at > all. Just the efforts of myself and several brave volunteers over the > years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > wrote: > Hearing that Asterisk (1.4) scales 2x like FS is not common, > sounds like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk > version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see > who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference > case, and > > it is the same results as for 1.0.4. The audio failed at > around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead > of 300%), > > and yet the audio still failed at the same number of > listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with > them and > > many of them are unfounded and take up a lot of time to > track down. > > That does not mean you have not found a real problem but the > first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > > wrote: > > > > I didn?t realize there was a policy about load testing > questions. What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I > do I will > > provide you with the feedback when I do. Just let me know > what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old > release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may > as well > > take the time to try it now that you have opened a forum > questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and > verify > > that each one can hear the audio perfectly and in time with > what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load > test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot > because we have > > a policy against entertaining load testing questions but if > you like > > asterisk, by all means, use it, and good luck to you if > those numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. > Are there > > substantial fixes to mod_conference in the FreeSWITCH trunk > that might > > increase capacity for my scenario of one speaker and many > listeners? > > If I want to put this into a production environment, I would > need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, > and doing > > the same scenario was able to get 1 speaker and 600 > listeners on a > > single conference with no audio issues. The CPU at that > point was just > > over 300%, same as where the single conference scenario > failed on > > FreeSWITCH with 300 listeners. I was able to push it to > over 700 > > listeners before I reached 400% CPU usage (I guess maxing > out my > > quad-core processors), and asterisk finally crashed. But up > until that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more > scalable > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more > than > > doubling FreeSWITCH capacity in this case. Again, maybe > there is > > something on the FreeSWITCH side that I?m doing wrong, but I > don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn > trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other > solutions. My > > scenario is to have one speaker, and many listeners (mute). > Since I > > have only one speaker, I was expecting this to scale well > because > > there is no audio mixing required, just send each frame of > the single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on > what I?ve > > read on how FreeSWITCH is supposed to be generally very > scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon > server, 4 Gig > > of RAM. I?ve set file logging to ?notice? level. My > conference profile > > is configured to suppress several events, hoping that it > would improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I > reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 > listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx > 110 > > listeners per conference (so just over 400 total channels on > the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the > system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH > process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar > testing, or > > maybe has suggestions on how to improve the performance. Or > perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users-list at metik.com Fri Dec 18 09:45:12 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 18 Dec 2009 12:45:12 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2B2188.2060803@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> Message-ID: <4B2BBFA8.9050900@metik.com> Honestly, several years ago I accomplished this by mod'ing SER (which became OpenSER which was then forked to OpenSIPS and Kamalio) and using one cluster of proxies for subscriber endpoints and another for infrastructure (so that I could keep RTP flows optimized yet support double NAT when required by an endpoint). Although the network looks different today. However, we were never quite happy about the lack of media failover (complicated NAT) and evaluated several commercial solutions until finding Covergence (which is now, for better or for worse since the jury is still out, owned by ACME Packet). At the time, they offered the best mix of security (their forte) yet scaled very well in comparison to their competitors that I had tested in our lab (ACME Packet, Kagoor, Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great decision, not unlike that of the FS developers, to implement a proven/stable SIP protocol stack. Nothing is perfect and that does not mean that we did not spend a considerable amount of time documenting bugs so that they could be addressed and it would work as it should We still use OpenSIPS for certain CSCF functionality (due to its speed and flexibility since it is not a B2BUA). Based on Mathieu's response (and he is definitely someone that would know), it looks like you should be able to easily append a X-AUTH-IP header (via OpenSIPS) containing the IP address of the endpoint and call it a day. -metik Bill W wrote: > Hey Metik, > > That's exactly what I'm trying to do... load balance across multiple FS > boxes, and have any machine in the cluster be able to reach a device > behind a NAT firewall. Hence the need for the proxy. Also, I'm trying > to keep the proxy relatively "dumb" and put all the logic in the FS boxes. > > True I could do the auth on the proxies as well, but then I'm setting up > another authentication scheme in addition to what is on the FS boxes, > and then integrating the databases so everything is consistent. > > I also have hosts that talk to the FS boxes directly, rather than > through the proxy. So I can't get rid of auth_acl on FS either, even if > I do implement it on the proxies. So my setup becomes much more > complex and potentially brittle. > > And all we're really talking about for FreeSWITCH, conceptually > speaking, is populating a variable with a different IP. We could even > make it configurable, as to which IP is to be used for the auth-acl. > > What are you using for SBCs? (if you are allowed to divulge that) I'm > currently using OpenSIPS for my proxy. > > Thanks, > Bill > > Metik wrote: > >> Why not simply implement this feature in the PROXY itself? >> >> FS has a pretty comprehensive security feature set for endpoints that >> directly register with it. >> >> Don't get me wrong, I do agree this is useful especially if you are >> going to be using your proxies to load balance across multiple FS boxes >> to create an ad-hoc cluster. I actually have session border controllers >> that have this feature and use it quite often. >> >> -metik >> >> Bill W wrote: >> >>> Hey Metik, >>> >>> Thanks for the reply, and the pointers for doing it with xml_curl. >>> >>> I'll guess have to do that in the short term, but in my opinion, having >>> auth-acl be able to work through a proxy is very important as it is a >>> vital part of a comprehensive security feature set. And it would be >>> much simpler to implement from an end-user perspective than the >>> alternative of doing it in xml_curl. >>> >>> As a matter of fact, I'm considering offering a bounty for that feature. >>> What is the going rate for that kind of thing? >>> >>> Is anyone out there interested in coding this feature? Or chipping in >>> for the bounty? >>> >>> >>> Thanks, >>> Bill >>> >>> >>> Metik wrote: >>> >>> >>>> This may be difficult considering that ACL needs to consider the >>>> original src IP/URI. To do that it, freeswitch would need to do so >>>> using a header that retains that information (i.e. From, Via, Contact, >>>> etc.). Which I do not believe is currently possible using auth-acl or >>>> apply-proxy-acl. >>>> >>>> However, you should be able to emulate the behavior using mod_xml_curl >>>> (and validating against appropriate variables available when using it to >>>> authenticate the request). >>>> >>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>> >>>> -metik >>>> >>>> >>>> Bill W wrote: >>>> >>>> >>>>> Hey Brian, >>>>> >>>>> >>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>> through a proxy the way I want them to, even with setting >>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>> >>>>> I have a multi-tenant system with multiple domains with multiple users >>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>> >>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>> the proxy. >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> Brian West wrote: >>>>> >>>>> >>>>> >>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>> >>>>>> /b >>>>>> >>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Okay, I added: to my sofia >>>>>>> profile and restarted sofia, and still no joy. >>>>>>> >>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>> I've got in >>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>> >>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>> >>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>> >>>>>>> Thoughts? >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>>> >>>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Dec 18 09:47:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 11:47:40 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <1261158104.11815.92.camel@luna.tc.commsmundi.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <1261158104.11815.92.camel@luna.tc.commsmundi.com> Message-ID: <191c3a030912180947s3e027aa8g712e1639934b6fe7@mail.gmail.com> yes, I understand. My reply was to the thread in general not directed at you =p On Fri, Dec 18, 2009 at 11:41 AM, Fran?ois Delawarde < fdelawarde at wirelessmundi.com> wrote: > It was of course just bad humor, I love both projects for what they are, > and I agree that both have their own advantages and inconvenients. > > For example, accessing that same conference from a dahdi card could be > another goal where Asterisk would be at an advantage, as chan_dahdi is > still superior (in the more tested sense) than openzap+mod_openzap. > > I just use both projects separately or together depending on what's > needed! > > I'm no banker nor do I understand the code, but many thanks for all > those unpaid contributions providing an excellent alternative for free > telephony. Your names really deserve being engraved in google's cache > for eternity. :-) > > But still, I would like to see those numbers... > > Fran?ois. > > > On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote: > > Conferencing is hardly the best place to judge performance. > > Quality is a far more important goal to me in conferencing. > > > > Lets compare who can do 48khz conferences with several 32k siren > > callers on a polycom 6000, several more using G722 at 16khz and > > another handful of people on g711 ulaw all at different rates and > > ptimes talking in near-real time with low delay and low echo. The > > fact that you can broadcast the conferences to icecast, control it > > from an external application and play files etc, and oh yeah, it can > > stream video. > > > > Frankly, considering this is a free software project and so many > > people benefit, i would rather focus on quality than what numbers i > > can get from having robots call the conference in some way that > > probably does not match reality. I would love for someone to sponsor > > the effort to add features to the conference module, but of course, I > > do not hold my breath, instead I continue to improve it for free when > > I find time. This is one of many reasons I do not enjoy performance > > discussions unless I am talking to an engineer who understands the > > code or a banker ready to pay for improvements. That is not my way of > > saying pay me or forget it as you can clearly see the conference > > module has made it to where it is today with no financial support at > > all. Just the efforts of myself and several brave volunteers over the > > years who have contributed to it. > > > > BTW, > > > > We have a weekly call, there is one today in 30 minutes. > > Drop by sip:888 at conference.freeswitch.orgThis is just an openVZ > > instance mind you running at 48khz waiting for anyone to call in and > > say hi. > > > > > > > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > > wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, > > sounds like > > a configuration error. > > > > If not, I already see the title of the next Digium blog entry: > > "FreeSwitch scalability myth finally ends: The worst Asterisk > > version > > ever (1.4) beating the crap of the best and latest FS." > > > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see > > who wins > > the final conference battle! :-) > > > > Fran?ois. > > > > > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > > I did a test with the trunk version for the one conference > > case, and > > > it is the same results as for 1.0.4. The audio failed at > > around 300 > > > listeners. Oddly though, it consumed less %CPU (240% instead > > of 300%), > > > and yet the audio still failed at the same number of > > listeners. > > > > > > > > > > > > Brian. > > > > > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > > Sent: Thursday, December 17, 2009 3:49 PM > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with > > them and > > > many of them are unfounded and take up a lot of time to > > track down. > > > That does not mean you have not found a real problem but the > > first > > > step is trying trunk. > > > > > > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > > > > wrote: > > > > > > I didn?t realize there was a policy about load testing > > questions. What > > > forum should I have used for this? > > > > > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I > > do I will > > > provide you with the feedback when I do. Just let me know > > what forum > > > to use for this topic from now on. > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > One man's stable release is another man's 6 month old > > release with > > > hundreds of known fixed bugs. > > > If one of the core developers tells you to try it, you may > > as well > > > take the time to try it now that you have opened a forum > > questioning > > > the scalability. > > > > > > When you tested asterisk did you actually use 600 phones and > > verify > > > that each one can hear the audio perfectly and in time with > > what the > > > speaker was saying? Did you try same on FS? > > > > > > Did you optimize your dialplan on FS to deal with a load > > test or > > > follow any of the recommended performance tuning page. > > > > > > All of the answers to these questions are really moot > > because we have > > > a policy against entertaining load testing questions but if > > you like > > > asterisk, by all means, use it, and good luck to you if > > those numbers > > > you are testing at are what you plan to put in real > > > production......... > > > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > > > > wrote: > > > > > > Hi Mike, > > > > > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. > > Are there > > > substantial fixes to mod_conference in the FreeSWITCH trunk > > that might > > > increase capacity for my scenario of one speaker and many > > listeners? > > > If I want to put this into a production environment, I would > > need a > > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, > > and doing > > > the same scenario was able to get 1 speaker and 600 > > listeners on a > > > single conference with no audio issues. The CPU at that > > point was just > > > over 300%, same as where the single conference scenario > > failed on > > > FreeSWITCH with 300 listeners. I was able to push it to > > over 700 > > > listeners before I reached 400% CPU usage (I guess maxing > > out my > > > quad-core processors), and asterisk finally crashed. But up > > until that > > > point, there were no audio problems. > > > > > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more > > scalable > > > than Asterisk, but unless there is something wrong with my > > FreeSWITCH > > > setup, Asterisk was clearly the winner in this test ? more > > than > > > doubling FreeSWITCH capacity in this case. Again, maybe > > there is > > > something on the FreeSWITCH side that I?m doing wrong, but I > > don?t see > > > what it could be. > > > > > > > > > > > > Brian. > > > > > > > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > > Sent: Thursday, December 17, 2009 10:18 AM > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > > > > > > I would be curious what the same tests produce with svn > > trunk of > > > FreeSWITCH. > > > > > > > > > > > > > > > Mike > > > > > > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > > > > > > Hi, > > > > > > > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > > mod_conference to see if it will scale better that other > > solutions. My > > > scenario is to have one speaker, and many listeners (mute). > > Since I > > > have only one speaker, I was expecting this to scale well > > because > > > there is no audio mixing required, just send each frame of > > the single > > > speaker to each listener. Unfortunately, my testing was > > disappointing, > > > and it didn?t scale nearly as well as I?d hoped (based on > > what I?ve > > > read on how FreeSWITCH is supposed to be generally very > > scalable). > > > > > > > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon > > server, 4 Gig > > > of RAM. I?ve set file logging to ?notice? level. My > > conference profile > > > is configured to suppress several events, hoping that it > > would improve > > > performance. > > > > > > > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I > > reached the > > > point of audio failure on the conferences: > > > > > > > > > > > > > > > > > > Scenario 1: > > > > > > > > > 1 conference, 1 speaker, audio failed at approx 300 > > listeners (mute) > > > > > > > > > > > > > > > > > > Scenario 2: > > > > > > > > > 4 conferences, 1 speaker per conference, audio failed approx > > 110 > > > listeners per conference (so just over 400 total channels on > > the > > > system). > > > > > > > > > > > > > > > > > > Scenario 3: > > > > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > > listeners > > > per conference (so just over 500 total channels on the > > system). > > > > > > > > > > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > > scenarios, > > > the audio quality failed when the % CPU for the FreeSWITCH > > process > > > exceeded 300%. > > > > > > > > > > > > > > > > > > I was hoping maybe someone else might have done similar > > testing, or > > > maybe has suggestions on how to improve the performance. Or > > perhaps an > > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > > > > > > > Thanks, > > > > > > > > > > > > > > > > > > Brian. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/4394927e/attachment-0002.html From djbinter at yahoo.com Fri Dec 18 10:09:29 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 10:09:29 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <156715.67103.qm@web37503.mail.mud.yahoo.com> Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/70548a5c/attachment-0002.html From msc at freeswitch.org Fri Dec 18 10:18:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 10:18:33 -0800 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> Message-ID: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij wrote: > Hi guys (and girls)! > > I'm working on a little bit of ENUM trickery and I tried doing some > (illegal) nested conditions. :-) > > What I want to do is to first check enum with the ENUM application, > then depending on the answer do some stuff. Say that the domain part > of the ENUM answer is robin.nl, then I want to do action X instead of > just briding the enum answer directly as I see in most examples. > > But I remembered that it wasn't allowed to do nested conditions. So > what I did was stacked conditions. After that I read the dialplan wiki > pages again and figured that my regexp never matches because variables > I "set" during some phase of the extension I can't use in the same > "go" as another condition. So, now my plan is to use LUA to do the > regexp. > > I'll pass the enum answer to a lua script which will split the answer > in a user and domain part and return those two to the main app. Then > based on those two vars I'll do routing or other actions (like, prefix > and then route). > > Is this how I'm supposed to do it? I can't find many examples on > manipulating ENUM answers, other than bridging them directly. I can't > change the way I do stuff to ENUM answers, because in most cases I'll > just route them out the standard way. > > Anyone with experience on fiddling with ENUM answers? > One thing you can do is create an extension that does the enum look up and then transfers the call back into the dialplan. You could set up a separate context that handles just the enum checking. Your condition would just need to match whatever var you put the enum return val in. So if your var name is "enum_res" then you can transfer like this after your enum lookup: Then create a context named "my_enum_context" and match for the condition(s) you need, like: do stuff Then have a different extension for other values of enum_res... This is just one way to do it without using a scripting lang. you can by all means use Lua as well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/a971d7d1/attachment-0002.html From djbinter at yahoo.com Fri Dec 18 10:23:18 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 10:23:18 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <618888.32098.qm@web37504.mail.mud.yahoo.com> Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/c0704cd0/attachment-0002.html From jerry.richards at teotech.com Fri Dec 18 10:35:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 18 Dec 2009 10:35:41 -0800 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header Message-ID: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry From freeswitch-users-list at metik.com Fri Dec 18 11:13:03 2009 From: freeswitch-users-list at metik.com (Metik) Date: Fri, 18 Dec 2009 14:13:03 -0500 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> Message-ID: <4B2BD43F.50903@metik.com> I use a similar method (transfer to XML dialplan based on the value of "${enum_route_1}") to determine if the SIP URI is native to a particular FS instance or if it needs to be sent off-net and it works well. -metik Michael Collins wrote: > > > On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij > wrote: > > Hi guys (and girls)! > > I'm working on a little bit of ENUM trickery and I tried doing some > (illegal) nested conditions. :-) > > What I want to do is to first check enum with the ENUM application, > then depending on the answer do some stuff. Say that the domain part > of the ENUM answer is robin.nl , then I want to > do action X instead of > just briding the enum answer directly as I see in most examples. > > But I remembered that it wasn't allowed to do nested conditions. So > what I did was stacked conditions. After that I read the dialplan wiki > pages again and figured that my regexp never matches because variables > I "set" during some phase of the extension I can't use in the same > "go" as another condition. So, now my plan is to use LUA to do the > regexp. > > I'll pass the enum answer to a lua script which will split the answer > in a user and domain part and return those two to the main app. Then > based on those two vars I'll do routing or other actions (like, prefix > and then route). > > Is this how I'm supposed to do it? I can't find many examples on > manipulating ENUM answers, other than bridging them directly. I can't > change the way I do stuff to ENUM answers, because in most cases I'll > just route them out the standard way. > > Anyone with experience on fiddling with ENUM answers? > > > One thing you can do is create an extension that does the enum look up > and then transfers the call back into the dialplan. You could set up a > separate context that handles just the enum checking. Your condition > would just need to match whatever var you put the enum return val in. > So if your var name is "enum_res" then you can transfer like this > after your enum lookup: > > > > Then create a context named "my_enum_context" and match for the > condition(s) you need, like: > > do stuff > > > Then have a different extension for other values of enum_res... > > This is just one way to do it without using a scripting lang. you can > by all means use Lua as well. > -MC > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at proximosystems.com Fri Dec 18 11:14:33 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 14:14:33 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> Message-ID: <00b901ca8016$55289800$ff79c800$@com> I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. My scenario is not a hypothetical one of ?having robots call the conference in a way that probably does not match reality?. In fact, this will very much reflect the reality of the application I?m building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum ? per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I?m trying to find a real solution to a real problem. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, December 18, 2009 11:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability Conferencing is hardly the best place to judge performance. Quality is a far more important goal to me in conferencing. Lets compare who can do 48khz conferences with several 32k siren callers on a polycom 6000, several more using G722 at 16khz and another handful of people on g711 ulaw all at different rates and ptimes talking in near-real time with low delay and low echo. The fact that you can broadcast the conferences to icecast, control it from an external application and play files etc, and oh yeah, it can stream video. Frankly, considering this is a free software project and so many people benefit, i would rather focus on quality than what numbers i can get from having robots call the conference in some way that probably does not match reality. I would love for someone to sponsor the effort to add features to the conference module, but of course, I do not hold my breath, instead I continue to improve it for free when I find time. This is one of many reasons I do not enjoy performance discussions unless I am talking to an engineer who understands the code or a banker ready to pay for improvements. That is not my way of saying pay me or forget it as you can clearly see the conference module has made it to where it is today with no financial support at all. Just the efforts of myself and several brave volunteers over the years who have contributed to it. BTW, We have a weekly call, there is one today in 30 minutes. Drop by sip:888 at conference.freeswitch.org This is just an openVZ instance mind you running at 48khz waiting for anyone to call in and say hi. On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde wrote: Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds like a configuration error. If not, I already see the title of the next Digium blog entry: "FreeSwitch scalability myth finally ends: The worst Asterisk version ever (1.4) beating the crap of the best and latest FS." Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins the final conference battle! :-) Fran?ois. On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > I did a test with the trunk version for the one conference case, and > it is the same results as for 1.0.4. The audio failed at around 300 > listeners. Oddly though, it consumed less %CPU (240% instead of 300%), > and yet the audio still failed at the same number of listeners. > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 3:49 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > We didn't post it anywhere but we just get overwhelmed with them and > many of them are unfounded and take up a lot of time to track down. > That does not mean you have not found a real problem but the first > step is trying trunk. > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > wrote: > > I didn?t realize there was a policy about load testing questions. What > forum should I have used for this? > > > > I didn?t get the chance to test on FS trunk yet, but when I do I will > provide you with the feedback when I do. Just let me know what forum > to use for this topic from now on. > > > > Thanks, > > > > Brian. > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Thursday, December 17, 2009 2:42 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > One man's stable release is another man's 6 month old release with > hundreds of known fixed bugs. > If one of the core developers tells you to try it, you may as well > take the time to try it now that you have opened a forum questioning > the scalability. > > When you tested asterisk did you actually use 600 phones and verify > that each one can hear the audio perfectly and in time with what the > speaker was saying? Did you try same on FS? > > Did you optimize your dialplan on FS to deal with a load test or > follow any of the recommended performance tuning page. > > All of the answers to these questions are really moot because we have > a policy against entertaining load testing questions but if you like > asterisk, by all means, use it, and good luck to you if those numbers > you are testing at are what you plan to put in real > production......... > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > wrote: > > Hi Mike, > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are there > substantial fixes to mod_conference in the FreeSWITCH trunk that might > increase capacity for my scenario of one speaker and many listeners? > If I want to put this into a production environment, I would need a > stable version, which as far as I know is the 1.0.4 version. > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > the same scenario was able to get 1 speaker and 600 listeners on a > single conference with no audio issues. The CPU at that point was just > over 300%, same as where the single conference scenario failed on > FreeSWITCH with 300 listeners. I was able to push it to over 700 > listeners before I reached 400% CPU usage (I guess maxing out my > quad-core processors), and asterisk finally crashed. But up until that > point, there were no audio problems. > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > than Asterisk, but unless there is something wrong with my FreeSWITCH > setup, Asterisk was clearly the winner in this test ? more than > doubling FreeSWITCH capacity in this case. Again, maybe there is > something on the FreeSWITCH side that I?m doing wrong, but I don?t see > what it could be. > > > > Brian. > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Thursday, December 17, 2009 10:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > I would be curious what the same tests produce with svn trunk of > FreeSWITCH. > > > > > Mike > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > Hi, > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > mod_conference to see if it will scale better that other solutions. My > scenario is to have one speaker, and many listeners (mute). Since I > have only one speaker, I was expecting this to scale well because > there is no audio mixing required, just send each frame of the single > speaker to each listener. Unfortunately, my testing was disappointing, > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > Here?s my server setup is this: > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig > of RAM. I?ve set file logging to ?notice? level. My conference profile > is configured to suppress several events, hoping that it would improve > performance. > > > > > > Here are a few scenarios I tested, and roughly where I reached the > point of audio failure on the conferences: > > > > > > Scenario 1: > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > Scenario 2: > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > listeners per conference (so just over 400 total channels on the > system). > > > > > > Scenario 3: > > > 16 conferences, 1 speaker per conference, audio failed at 32 listeners > per conference (so just over 500 total channels on the system). > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > the audio quality failed when the % CPU for the FreeSWITCH process > exceeded 300%. > > > > > > I was hoping maybe someone else might have done similar testing, or > maybe has suggestions on how to improve the performance. Or perhaps an > alternate solution to the one speaker, many listener case? > > > > > > Thanks, > > > > > > Brian. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/a880f3d8/attachment-0002.html From msc at freeswitch.org Fri Dec 18 11:33:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Dec 2009 11:33:02 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > I was evaluating the technologies available, and I thought you would be > interested in my results. However, almost every other reply I get from you > to my posts, rather than being helpful, has been hostile and insulting. > Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) > > > My scenario is not a hypothetical one of ?having robots call the conference > in a way that probably does not match reality?. In fact, this will very much > reflect the reality of the application I?m building. Only instead of 300 > listeners, I need to scale to over 2000 listeners minimum ? per event, with > possibly more than one concurrent event. I want to pack as many listeners on > one server as I can. I?m trying to find a real solution to a real problem. > That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. > > > I work with other open source projects and fund enhancements or fixes I > need. FreeSWITCH would be no different. > > > Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/792c7412/attachment-0002.html From lon at kickasspixels.com Fri Dec 18 11:41:12 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 18 Dec 2009 11:41:12 -0800 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <2B659DC6-32F0-4046-93E5-22E3E5EEEF65@kickasspixels.com> Brian, Now that you know the scale freeswotch scales to in you scenario, and having designed a mult-server solution can you not add more server to scale further? As freeswitch continues to improve retest and revise your architecture design. Sent from my iPhone On Dec 18, 2009, at 11:14 AM, Brian wrote: > I was evaluating the technologies available, and I thought you would > be interested in my results. However, almost every other reply I get > from you to my posts, rather than being helpful, has been hostile > and insulting. > > > > My scenario is not a hypothetical one of ?having robots call the con > ference in a way that probably does not match reality?. In fact, thi > s will very much reflect the reality of the application I?m building > . Only instead of 300 listeners, I need to scale to over 2000 listen > ers minimum ? per event, with possibly more than one concurrent even > t. I want to pack as many listeners on one server as I can. I?m tryi > ng to find a real solution to a real problem. > > > > I work with other open source projects and fund enhancements or > fixes I need. FreeSWITCH would be no different. > > > > Brian. > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, December 18, 2009 11:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to > sponsor the effort to add features to the conference module, but of > course, I do not hold my breath, instead I continue to improve it > for free when I find time. This is one of many reasons I do not > enjoy performance discussions unless I am talking to an engineer who > understands the code or a banker ready to pay for improvements. > That is not my way of saying pay me or forget it as you can clearly > see the conference module has made it to where it is today with no > financial support at all. Just the efforts of myself and several > brave volunteers over the years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde m> wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds > like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. > What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I w > ill > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we > have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those > numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are th > ere > > substantial fixes to mod_conference in the FreeSWITCH trunk that > might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was > just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until > that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalab > le > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I > don?t see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other > solutions. My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the > single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what > I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 > Gig > > of RAM. I?ve set file logging to ?notice? level. My > conference profile > > is configured to suppress several events, hoping that it would > improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 > scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or > perhaps an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/31d920c9/attachment-0002.html From viper at fx-services.com Fri Dec 18 11:51:37 2009 From: viper at fx-services.com (Robin Vleij) Date: Fri, 18 Dec 2009 20:51:37 +0100 Subject: [Freeswitch-users] LUA and return variables In-Reply-To: <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> References: <20091218112112.15771h028hr5dwns@monsoon.fx-services.com> <87f2f3b90912181018s3bf87189w3f606b1525ab7039@mail.gmail.com> Message-ID: <4B2BDD49.9030908@fx-services.com> On 12/18/09 7:18 PM, Michael Collins wrote: Hi Michael, > One thing you can do is create an extension that does the enum look up > and then transfers the call back into the dialplan. You could set up a Cool idea, didn't think about that! > separate context that handles just the enum checking. Your condition > would just need to match whatever var you put the enum return val in. So > if your var name is "enum_res" then you can transfer like this after > your enum lookup: Right, makes sense. Going to try a bit in that direction. Do the enum lookup and then transfer to an enum handling context. Simple, should have thought about that. :) > This is just one way to do it without using a scripting lang. you can by > all means use Lua as well. My main question there really was, since I'm not able to work on vars I set in an extention, will that work if I return vars from a script? It should really, but I was asking to make sure it would. I think in that design, the script would have been like three rules or something, but keeping it in the dialplan is nicer, I think (even though it says "don't do magic in the dialplan, do it in scripts" on the wiki). I'll report back when I managed to fiddle something together. /Robin From anthony.minessale at gmail.com Fri Dec 18 11:55:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 13:55:38 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> Message-ID: <191c3a030912181155h411877c9y9b9d40849f54dd0@mail.gmail.com> Brian, there was not one insulting word in anything I have said and as this is a community mailing list my replies are always voiced to address the public in general not you specifically, like I already mentioned in my last post. If you open a public forum on a FAQ be prepared to hear our policy. Indeed many people do unrealistic load testing and most people with strong will find it insulting when a group of people have a set of standard policy by which they try to deal with making a penny jar for all the 2 cents worth of input we get on a daily basis. I can't begin to iterate over all the cases we endure on a weekly basis. additionally 90% of bug reports are on older releases and we always make people reproduce their issues on SVN trunk because 3 core devs and a handful of helpers can't maintain 20 versions of the code. I gave you some really suggestions yesterday let me repaste it, I fail to see any insults: --------------------------------------------------------------------------------------------------------------------------------------------------- What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well. like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp --------------------------------------------------------------------------------------------------------------------------------------------------- I have to get in these "fights" with people constantly so I guess that is part of my job and my biggest mistake is spending so much time trying to explain myself. - Show quoted text - On Fri, Dec 18, 2009 at 1:33 PM, Michael Collins wrote: > > > On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > >> I was evaluating the technologies available, and I thought you would be >> interested in my results. However, almost every other reply I get from you >> to my posts, rather than being helpful, has been hostile and insulting. >> > Thanks for your input. Just so you know, Tony deals with people on a near > daily basis who want to spend time doing crazy schemes under the guise of > "load testing" or "researching a new solution" which are not grounded in > reality. At first blush this scenario sounded like one of those schemes. > However it definitely looks like you've built a test scenario that mimics > reality better than most. I think we can give you a pass for not being able > to get 500 people all at once to call in every time you need to test. :) > >> >> >> My scenario is not a hypothetical one of ?having robots call the >> conference in a way that probably does not match reality?. In fact, this >> will very much reflect the reality of the application I?m building. Only >> instead of 300 listeners, I need to scale to over 2000 listeners minimum ? >> per event, with possibly more than one concurrent event. I want to pack as >> many listeners on one server as I can. I?m trying to find a real solution to >> a real problem. >> > That kind of volume suggests that the icecast style solution would be best. > It takes much less resources to send audio in one direction than it does to > mix audio from multiple parties. I like bkw's initial suggestion of > transferring a caller to the conference only when he/she needs to speak, > such as to ask a question. Like Tony mentioned, his focus is on quality not > quantity, so mod_conference probably isn't the best tool for this scenario. > >> >> >> I work with other open source projects and fund enhancements or fixes I >> need. FreeSWITCH would be no different. >> >> >> > Excellent! It looks like we don't already have a canned solution, > obviously, but as bkw likes to say, all the Lego bricks are there to build > the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the > weekly conference which is going on right now and you might catch some of > the devs and leading community members and you can chat in real-time about > your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) > > -Michael > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/35731e70/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 18 12:02:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:02:14 -0600 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header In-Reply-To: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> References: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Message-ID: <191c3a030912181202h471fa07n3f974664b64b6b1b@mail.gmail.com> could be possible with a code change, open a bounty on jira and someone may do it On Fri, Dec 18, 2009 at 12:35 PM, Jerry Richards wrote: > Is it possible to allow/deny REGISTER requests based on the User-Agent > header? I need to know/manage what devices are registering. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/38fb823a/attachment-0002.html From yehavi.bourvine at gmail.com Fri Dec 18 12:16:04 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 18 Dec 2009 22:16:04 +0200 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <26842055.post@talk.nabble.com> References: <26842055.post@talk.nabble.com> Message-ID: Try the following: I don't know whether it will work in your case, but here we use it to reject a call while we want to signal that the remote party is busy. Regards, __Yehavi: 2009/12/18 bcxml > > I have an incomming call being answered by FreeSwitch and passed to IVR > application which rejects the call. > > The call is never answered by FreeSwitch, but instead of hearing a busy > signal, the caller hears ringing. > > Can anyone advise how I can get the user to hear a busy signal after call > rejection instead of ringing. > > Here is the debug trace > > http://pastebin.freeswitch.org/11558 > > Thanks > > > Brian > > -- > View this message in context: > http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/4eb2b2ff/attachment-0002.html From brian at proximosystems.com Fri Dec 18 12:16:28 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 15:16:28 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> Message-ID: <00d801ca801e$fb7d5670$f2780350$@com> Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/5e61f879/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 18 12:30:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:30:29 -0600 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00d801ca801e$fb7d5670$f2780350$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> <00d801ca801e$fb7d5670$f2780350$@com> Message-ID: <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consulting at freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian wrote: > Hi Michael, > > > > Thanks for the invite, but I can?t make it on the call. Anyway, I?m not > sure if discussing my specific case is meant for that type of call, is it? > > > > After Brian?s suggestion to use shoutcast and local streams, I was looking > at the code for those modules. I?m not familiar with shoutcast or icecast > capabilities, so I don?t know if they can just pass though my audio stream > unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on > the source server, and then back from mp3 to uLaw (or whatever phone codec) > on the other server. > > > > I was wondering if maybe there was a way to make a stream out of an > existing channel, and have all the other channels just listen to that > stream. It would be sort of halfway between conference and shoutcast. I > would call in to the secondary server like I already do, but only instead of > entering into a conference as a speaker, the channel would just start > producing a local audio stream for the listener channels to tap into. It > would avoid the need to have another piece of software to manage (shoutcast > or icecast), and my support team would be happier... > > > > However, I would still need to do tests for the streaming idea to see how > that scales... > > > > Brian. > > > > > > *From:* Michael Collins [mailto:msc at freeswitch.org] > *Sent:* Friday, December 18, 2009 2:33 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_conference scalability > > > > > > On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: > > I was evaluating the technologies available, and I thought you would be > interested in my results. However, almost every other reply I get from you > to my posts, rather than being helpful, has been hostile and insulting. > > Thanks for your input. Just so you know, Tony deals with people on a near > daily basis who want to spend time doing crazy schemes under the guise of > "load testing" or "researching a new solution" which are not grounded in > reality. At first blush this scenario sounded like one of those schemes. > However it definitely looks like you've built a test scenario that mimics > reality better than most. I think we can give you a pass for not being able > to get 500 people all at once to call in every time you need to test. :) > > > > My scenario is not a hypothetical one of ?having robots call the conference > in a way that probably does not match reality?. In fact, this will very much > reflect the reality of the application I?m building. Only instead of 300 > listeners, I need to scale to over 2000 listeners minimum ? per event, with > possibly more than one concurrent event. I want to pack as many listeners on > one server as I can. I?m trying to find a real solution to a real problem. > > That kind of volume suggests that the icecast style solution would be > best. It takes much less resources to send audio in one direction than it > does to mix audio from multiple parties. I like bkw's initial suggestion of > transferring a caller to the conference only when he/she needs to speak, > such as to ask a question. Like Tony mentioned, his focus is on quality not > quantity, so mod_conference probably isn't the best tool for this scenario. > > > > I work with other open source projects and fund enhancements or fixes I > need. FreeSWITCH would be no different. > > > > Excellent! It looks like we don't already have a canned solution, > obviously, but as bkw likes to say, all the Lego bricks are there to build > the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the > weekly conference which is going on right now and you might catch some of > the devs and leading community members and you can chat in real-time about > your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) > > -Michael > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/84eb348c/attachment-0002.html From anthony.minessale at gmail.com Fri Dec 18 12:31:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Dec 2009 14:31:28 -0600 Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: References: <26842055.post@talk.nabble.com> Message-ID: <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> that will only work if you have not answered yet. if you already have, you would need to indicate the tones inband like I mentioned. On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine wrote: > Try the following: > > > I don't know whether it will work in your case, but here we use it to > reject a call while we want to signal that the remote party is busy. > > Regards, __Yehavi: > > > > 2009/12/18 bcxml > > >> I have an incomming call being answered by FreeSwitch and passed to IVR >> application which rejects the call. >> >> The call is never answered by FreeSwitch, but instead of hearing a busy >> signal, the caller hears ringing. >> >> Can anyone advise how I can get the user to hear a busy signal after call >> rejection instead of ringing. >> >> Here is the debug trace >> >> http://pastebin.freeswitch.org/11558 >> >> Thanks >> >> >> Brian >> >> -- >> View this message in context: >> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/5afd7076/attachment-0002.html From dave at 3c.co.uk Fri Dec 18 12:56:41 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 18 Dec 2009 13:56:41 -0700 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <00b901ca8016$55289800$ff79c800$@com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <049601ca7f4f$37da5580$a78f0080$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> Message-ID: <1261169801.6033.3.camel@local.freepabx.com> Hi Brian, Have a look at this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop - I took a quick look through the code and couldn't see any reason why you shouldn't have a bunch of eavesdroppers listening to a single caller. I'd be surprised if this didn't perform a lot better for your application. Cheers -- Dave > I was evaluating the technologies available, and I thought you would > be interested in my results. However, almost every other reply I get > from you to my posts, rather than being helpful, has been hostile and > insulting. > > > > My scenario is not a hypothetical one of ?having robots call the > conference in a way that probably does not match reality?. In fact, > this will very much reflect the reality of the application I?m > building. Only instead of 300 listeners, I need to scale to over 2000 > listeners minimum ? per event, with possibly more than one concurrent > event. I want to pack as many listeners on one server as I can. I?m > trying to find a real solution to a real problem. > > > > I work with other open source projects and fund enhancements or fixes > I need. FreeSWITCH would be no different. > > > > Brian. > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, December 18, 2009 11:34 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > Conferencing is hardly the best place to judge performance. > Quality is a far more important goal to me in conferencing. > > Lets compare who can do 48khz conferences with several 32k siren > callers on a polycom 6000, several more using G722 at 16khz and > another handful of people on g711 ulaw all at different rates and > ptimes talking in near-real time with low delay and low echo. The > fact that you can broadcast the conferences to icecast, control it > from an external application and play files etc, and oh yeah, it can > stream video. > > Frankly, considering this is a free software project and so many > people benefit, i would rather focus on quality than what numbers i > can get from having robots call the conference in some way that > probably does not match reality. I would love for someone to sponsor > the effort to add features to the conference module, but of course, I > do not hold my breath, instead I continue to improve it for free when > I find time. This is one of many reasons I do not enjoy performance > discussions unless I am talking to an engineer who understands the > code or a banker ready to pay for improvements. That is not my way of > saying pay me or forget it as you can clearly see the conference > module has made it to where it is today with no financial support at > all. Just the efforts of myself and several brave volunteers over the > years who have contributed to it. > > BTW, > > We have a weekly call, there is one today in 30 minutes. > Drop by sip:888 at conference.freeswitch.org This is just an openVZ > instance mind you running at 48khz waiting for anyone to call in and > say hi. > > > > > > > On Fri, Dec 18, 2009 at 10:12 AM, Fran?ois Delawarde > wrote: > > Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds > like > a configuration error. > > If not, I already see the title of the next Digium blog entry: > "FreeSwitch scalability myth finally ends: The worst Asterisk version > ever (1.4) beating the crap of the best and latest FS." > > Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins > the final conference battle! :-) > > Fran?ois. > > > > On Thu, 2009-12-17 at 16:41 -0500, Brian wrote: > > I did a test with the trunk version for the one conference case, and > > it is the same results as for 1.0.4. The audio failed at around 300 > > listeners. Oddly though, it consumed less %CPU (240% instead of > 300%), > > and yet the audio still failed at the same number of listeners. > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 3:49 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > We didn't post it anywhere but we just get overwhelmed with them and > > many of them are unfounded and take up a lot of time to track down. > > That does not mean you have not found a real problem but the first > > step is trying trunk. > > > > > > > > > > On Thu, Dec 17, 2009 at 2:32 PM, Brian > > wrote: > > > > I didn?t realize there was a policy about load testing questions. > What > > forum should I have used for this? > > > > > > > > I didn?t get the chance to test on FS trunk yet, but when I do I > will > > provide you with the feedback when I do. Just let me know what forum > > to use for this topic from now on. > > > > > > > > Thanks, > > > > > > > > Brian. > > > > > > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Thursday, December 17, 2009 2:42 PM > > > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > One man's stable release is another man's 6 month old release with > > hundreds of known fixed bugs. > > If one of the core developers tells you to try it, you may as well > > take the time to try it now that you have opened a forum questioning > > the scalability. > > > > When you tested asterisk did you actually use 600 phones and verify > > that each one can hear the audio perfectly and in time with what the > > speaker was saying? Did you try same on FS? > > > > Did you optimize your dialplan on FS to deal with a load test or > > follow any of the recommended performance tuning page. > > > > All of the answers to these questions are really moot because we > have > > a policy against entertaining load testing questions but if you like > > asterisk, by all means, use it, and good luck to you if those > numbers > > you are testing at are what you plan to put in real > > production......... > > > > On Thu, Dec 17, 2009 at 1:29 PM, Brian > > wrote: > > > > Hi Mike, > > > > > > > > I didn?t get around to testing on the FreeSWITCH trunk yet. Are > there > > substantial fixes to mod_conference in the FreeSWITCH trunk that > might > > increase capacity for my scenario of one speaker and many listeners? > > If I want to put this into a production environment, I would need a > > stable version, which as far as I know is the 1.0.4 version. > > > > > > > > However, I did test on Asterisk 1.4 using app_conference, and doing > > the same scenario was able to get 1 speaker and 600 listeners on a > > single conference with no audio issues. The CPU at that point was > just > > over 300%, same as where the single conference scenario failed on > > FreeSWITCH with 300 listeners. I was able to push it to over 700 > > listeners before I reached 400% CPU usage (I guess maxing out my > > quad-core processors), and asterisk finally crashed. But up until > that > > point, there were no audio problems. > > > > > > > > I?ve read a lot about how FreeSWITCH is supposed to be more scalable > > than Asterisk, but unless there is something wrong with my > FreeSWITCH > > setup, Asterisk was clearly the winner in this test ? more than > > doubling FreeSWITCH capacity in this case. Again, maybe there is > > something on the FreeSWITCH side that I?m doing wrong, but I don?t > see > > what it could be. > > > > > > > > Brian. > > > > > > > > > > > > From: Michael Jerris [mailto:mike at jerris.com] > > Sent: Thursday, December 17, 2009 10:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] mod_conference scalability > > > > > > > > > > I would be curious what the same tests produce with svn trunk of > > FreeSWITCH. > > > > > > > > > > Mike > > > > > > > > > > On Dec 16, 2009, at 4:49 PM, Brian wrote: > > > > > > > > > > Hi, > > > > > > > > > > > > I?m new to FreeSWITCH and I?m testing the scalability of > > mod_conference to see if it will scale better that other solutions. > My > > scenario is to have one speaker, and many listeners (mute). Since I > > have only one speaker, I was expecting this to scale well because > > there is no audio mixing required, just send each frame of the > single > > speaker to each listener. Unfortunately, my testing was > disappointing, > > and it didn?t scale nearly as well as I?d hoped (based on what I?ve > > read on how FreeSWITCH is supposed to be generally very scalable). > > > > > > > > > > > > Here?s my server setup is this: > > > > > > > > > > > > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 > Gig > > of RAM. I?ve set file logging to ?notice? level. My conference > profile > > is configured to suppress several events, hoping that it would > improve > > performance. > > > > > > > > > > > > Here are a few scenarios I tested, and roughly where I reached the > > point of audio failure on the conferences: > > > > > > > > > > > > Scenario 1: > > > > > > 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) > > > > > > > > > > > > Scenario 2: > > > > > > 4 conferences, 1 speaker per conference, audio failed approx 110 > > listeners per conference (so just over 400 total channels on the > > system). > > > > > > > > > > > > Scenario 3: > > > > > > 16 conferences, 1 speaker per conference, audio failed at 32 > listeners > > per conference (so just over 500 total channels on the system). > > > > > > > > > > > > > > > > > > Looking at the output from ?top?, it seems that in all 3 scenarios, > > the audio quality failed when the % CPU for the FreeSWITCH process > > exceeded 300%. > > > > > > > > > > > > I was hoping maybe someone else might have done similar testing, or > > maybe has suggestions on how to improve the performance. Or perhaps > an > > alternate solution to the one speaker, many listener case? > > > > > > > > > > > > Thanks, > > > > > > > > > > > > Brian. > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From djbinter at yahoo.com Fri Dec 18 13:09:54 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 18 Dec 2009 13:09:54 -0800 (PST) Subject: [Freeswitch-users] SIP Re-invite In-Reply-To: <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> References: <607753.94827.qm@web37503.mail.mud.yahoo.com> <191c3a030912161542h2c2c32c5q12f4bf2c77d90f2f@mail.gmail.com> <206417.45016.qm@web37507.mail.mud.yahoo.com> <35773735-4CD8-417F-AAB5-9102E55C6CC0@jerris.com> <922386.16417.qm@web37502.mail.mud.yahoo.com> <7A2B359D-96E5-43F9-A223-8A9238F0561E@jerris.com> Message-ID: <707033.51380.qm@web37501.mail.mud.yahoo.com> Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that I thought it was a bug. Thank you again for the community and your support. Dorn B. ________________________________ From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, > > >My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 > > >Thank you, >Dorn B. > > ________________________________ From: Michael Jerris >To: freeswitch-users at lists.freeswitch.org >Sent: Thu, December 17, 2009 8:03:46 AM >Subject: Re: [Freeswitch-users] SIP Re-invite > >are you doing this trace from the freeswitch box itself? > > >Mike > > >On Dec 17, 2009, at 10:48 AM, DJB wrote: > >Anthony, >> >>I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 >>Please advise if you need further info. >> >>Thank you. >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/dfcb75e4/attachment-0002.html From brian at proximosystems.com Fri Dec 18 13:15:53 2009 From: brian at proximosystems.com (Brian) Date: Fri, 18 Dec 2009 16:15:53 -0500 Subject: [Freeswitch-users] mod_conference scalability In-Reply-To: <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> References: <00aa01ca7e99$9901f9a0$cb05ece0$@com> <191c3a030912171142j24e64cb9r1f811d76db8a4918@mail.gmail.com> <04b501ca7f58$0a188870$1e499950$@com> <191c3a030912171248y5cdef8cr6a2eb031f3608efd@mail.gmail.com> <04ca01ca7f61$a0560e30$e1022a90$@com> <1261152728.11815.57.camel@luna.tc.commsmundi.com> <191c3a030912180834q8537250oa212e5b81c5ae300@mail.gmail.com> <00b901ca8016$55289800$ff79c800$@com> <87f2f3b90912181133m6cf706a0nc819cfc92c985a41@mail.gmail.com> <00d801ca801e$fb7d5670$f2780350$@com> <191c3a030912181230p71d2e74av31ff5decd2307ca@mail.gmail.com> Message-ID: <010c01ca8027$48521820$d8f64860$@com> Sounds like a plan. We will pursue it through the consulting at freeswith.org route. Thanks, Brian. From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, December 18, 2009 3:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability I am more than sure there is probably plenty of room for conference optimizations it's just a big task. We don't have a test labbed up and an urgency to work on it. If you really want us to pursue trying to improve the performance perhaps you can contact us at consulting at freeswitch.org and provide us with access your test environment and let us investigate the possibility of making improvements. On Fri, Dec 18, 2009 at 2:16 PM, Brian wrote: Hi Michael, Thanks for the invite, but I can't make it on the call. Anyway, I'm not sure if discussing my specific case is meant for that type of call, is it? After Brian's suggestion to use shoutcast and local streams, I was looking at the code for those modules. I'm not familiar with shoutcast or icecast capabilities, so I don't know if they can just pass though my audio stream unchanged (as uLaw packets). I want to avoid converting from uLaw to mp3 on the source server, and then back from mp3 to uLaw (or whatever phone codec) on the other server. I was wondering if maybe there was a way to make a stream out of an existing channel, and have all the other channels just listen to that stream. It would be sort of halfway between conference and shoutcast. I would call in to the secondary server like I already do, but only instead of entering into a conference as a speaker, the channel would just start producing a local audio stream for the listener channels to tap into. It would avoid the need to have another piece of software to manage (shoutcast or icecast), and my support team would be happier... However, I would still need to do tests for the streaming idea to see how that scales... Brian. From: Michael Collins [mailto:msc at freeswitch.org] Sent: Friday, December 18, 2009 2:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_conference scalability On Fri, Dec 18, 2009 at 11:14 AM, Brian wrote: I was evaluating the technologies available, and I thought you would be interested in my results. However, almost every other reply I get from you to my posts, rather than being helpful, has been hostile and insulting. Thanks for your input. Just so you know, Tony deals with people on a near daily basis who want to spend time doing crazy schemes under the guise of "load testing" or "researching a new solution" which are not grounded in reality. At first blush this scenario sounded like one of those schemes. However it definitely looks like you've built a test scenario that mimics reality better than most. I think we can give you a pass for not being able to get 500 people all at once to call in every time you need to test. :) My scenario is not a hypothetical one of "having robots call the conference in a way that probably does not match reality". In fact, this will very much reflect the reality of the application I'm building. Only instead of 300 listeners, I need to scale to over 2000 listeners minimum - per event, with possibly more than one concurrent event. I want to pack as many listeners on one server as I can. I'm trying to find a real solution to a real problem. That kind of volume suggests that the icecast style solution would be best. It takes much less resources to send audio in one direction than it does to mix audio from multiple parties. I like bkw's initial suggestion of transferring a caller to the conference only when he/she needs to speak, such as to ask a question. Like Tony mentioned, his focus is on quality not quantity, so mod_conference probably isn't the best tool for this scenario. I work with other open source projects and fund enhancements or fixes I need. FreeSWITCH would be no different. Excellent! It looks like we don't already have a canned solution, obviously, but as bkw likes to say, all the Lego bricks are there to build the solution. Hop on IRC (#freeswitch in irc.freenode.net) or join the weekly conference which is going on right now and you might catch some of the devs and leading community members and you can chat in real-time about your challenges. (http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14) -Michael _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/335fdf8c/attachment-0002.html From bcxml at hotmail.com Fri Dec 18 14:16:56 2009 From: bcxml at hotmail.com (bcxml) Date: Fri, 18 Dec 2009 14:16:56 -0800 (PST) Subject: [Freeswitch-users] Ringing after call has been rejected In-Reply-To: <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> References: <26842055.post@talk.nabble.com> <191c3a030912181231k1cf38347g848288614e82f864@mail.gmail.com> Message-ID: <26850453.post@talk.nabble.com> Actually my application returns 403 if it decides that it doesnt want to answer the call So I changed the response that my applicatgion gives to 486 and I now get the behavior that I wanted Thanks for the advice Brian Anthony Minessale-2 wrote: > > that will only work if you have not answered yet. > if you already have, you would need to indicate the tones inband like I > mentioned. > > > On Fri, Dec 18, 2009 at 2:16 PM, Yehavi Bourvine > wrote: > >> Try the following: >> >> >> I don't know whether it will work in your case, but here we use it to >> reject a call while we want to signal that the remote party is busy. >> >> Regards, __Yehavi: >> >> >> >> 2009/12/18 bcxml >> >> >>> I have an incomming call being answered by FreeSwitch and passed to IVR >>> application which rejects the call. >>> >>> The call is never answered by FreeSwitch, but instead of hearing a busy >>> signal, the caller hears ringing. >>> >>> Can anyone advise how I can get the user to hear a busy signal after >>> call >>> rejection instead of ringing. >>> >>> Here is the debug trace >>> >>> http://pastebin.freeswitch.org/11558 >>> >>> Thanks >>> >>> >>> Brian >>> >>> -- >>> View this message in context: >>> http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26842055.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://old.nabble.com/Ringing-after-call-has-been-rejected-tp26842055p26850453.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From frank at carmickle.com Fri Dec 18 14:22:45 2009 From: frank at carmickle.com (Frank Carmickle) Date: Fri, 18 Dec 2009 17:22:45 -0500 Subject: [Freeswitch-users] packaging preference question Message-ID: <20091218222245.GE31924@base.carmickle.com> Hello all The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? --FC From quentusrex at gmail.com Fri Dec 18 14:40:32 2009 From: quentusrex at gmail.com (William King) Date: Fri, 18 Dec 2009 14:40:32 -0800 Subject: [Freeswitch-users] packaging preference question In-Reply-To: <20091218222245.GE31924@base.carmickle.com> References: <20091218222245.GE31924@base.carmickle.com> Message-ID: <1261176032.8965.14.camel@quentusrex-desktop> I think that sounds like a good idea. It would also keep permission management simple. -William King On Fri, 2009-12-18 at 17:22 -0500, Frank Carmickle wrote: > Hello all > > The packaging folk are interested in knowing if anyone has a problem with having the install set up the user and group to freeswitch:freeswitch. This would be the default on debs rpms and ports packaging. The freeswitch user would be added to daemon and audio groups. The FusionPBX packaging can then add www-data/apache to the freeswitch group. Any objections? > > --FC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at hijacked.us Fri Dec 18 17:43:59 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 18 Dec 2009 20:43:59 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) Message-ID: <20091219014359.GA21798@hijacked.us> So, it's been a while since I mentioned this project, but its finally nearing the point where it's going to be able to go into production (and replace my old asterisk-based platform) so I decided to dredge it up again. Briefly, spice telephony is a call/contact center platform that leverages FS for VoIP, IVRs, call recording, etc. It also supports handing email/voicemail contacts (chat is planned, too). Here's some features: * Skill based routing * Priority, unified queues * Web based administration, agent interaction (using the dojo toolkit) * Supervisory drag and drop interface for managing agents/call flow * Queue 'recipes' - ability to play announcements, send to voicemail, modify skills or priority based on certain conditions (queue time, media type, hour of day, # of available agents, etc). * Integration API for importing agents/clients out of a CRM/AD/whatever * Detailed CDRs recording every step of a call (IVR, Queue, Ring, Transfer, Wrapup, etc). The project is implemented in Erlang (erlang.org) and thus allows spice-telephony to be deployed as a distributed system (multiple nodes aggregated into a single system). Calls can come into any node and, skills permitting, can be offered to any agent on the local node or any of the remote nodes. Nodes can also operate independantly if isolated by a netsplit or simply deployed standalone. CDRs and config files are stored in erlang's distributed database, mnesia, and CDRs can be output in parallel to any node configured to do so (so you can have all your call data in multiple places without having to do SQL replication). Erlang's fault tolerant nature also allows the platform to be very robust, entire subsystems can fail at runtime and be automatically restarted by supervisor process, and the entire erlang node can be automatically restarted if the node crashes. There's a lot more than mentioned above, so I'd encourage anyone interested to grab the latest release from: http://opencsm.org/downloads/spice-telephony-0.9.6.tar.gz and look at the install guide: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony_Install_Guide You'll need an erlang version >= R13B01 and ruby's 'rake' installed, you shouldn't need much of anything else. It *does* work on windows but I don't recommend it (I can try to help you get it working though). There's also some more information available here: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony The documentation is a little sparse, but I'll do my best to answer any questions. Any feedback is appreciated. Andrew From andrew at hijacked.us Fri Dec 18 19:16:49 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 18 Dec 2009 22:16:49 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091219014359.GA21798@hijacked.us> References: <20091219014359.GA21798@hijacked.us> Message-ID: <20091219031649.GA1956@hijacked.us> I've been asked to provide some screenshots, so here's some of the agent/supervisor interface: http://eagle.bsd.st/~andrew/cpxshots/ Hopefully the image names are self-explanatory. In the ringing picture, that URL pop is a configurable URL that can be used to integrate with a CRM, in my case our own CRM - spicecsm. The URL supports interpolation for variables like callerid, clientid, call type, etc. The supervisor view is a little hard to describe via static images, but you're able to drag and drop agents into another profile (empty profiles are hidden when not dragging an agent), drag agents onto an agent to send them the call, and there's also various right click menus available. Oh, and I forgot to mention this before; this system is in 'live testing' and the goal is to do a final deployment sometime in January. Andrew From freeswitch at aastral.net Fri Dec 18 21:37:24 2009 From: freeswitch at aastral.net (Bill W.) Date: Sat, 19 Dec 2009 00:37:24 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2BBFA8.9050900@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> Message-ID: <4B2C6694.3060400@aastral.net> Hey Metik, Thanks so much for your insights and your help. And yes, I was able to append the X-AUTH-IP header with no problem. But that didn't solve the issue. After some more research, it appears that the problem isn't with auth-calls at all. I disabled all auth-call directives in every sip profile and the registration through a proxy is still being rejected. I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the ip variable against the auth_acl cidr. if (auth_acl) { if (!switch_check_network_list_ip(ip, auth_acl)) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); ret = AUTH_FORBIDDEN; goto end; } So I guess the question is, is it possible to control what gets put into the ip variable? Thanks, Bill Metik wrote: > Honestly, several years ago I accomplished this by mod'ing SER (which > became OpenSER which was then forked to OpenSIPS and Kamalio) and using > one cluster of proxies for subscriber endpoints and another for > infrastructure (so that I could keep RTP flows optimized yet support > double NAT when required by an endpoint). Although the network looks > different today. > > However, we were never quite happy about the lack of media failover > (complicated NAT) and evaluated several commercial solutions until > finding Covergence (which is now, for better or for worse since the jury > is still out, owned by ACME Packet). At the time, they offered the best > mix of security (their forte) yet scaled very well in comparison to > their competitors that I had tested in our lab (ACME Packet, Kagoor, > Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great > decision, not unlike that of the FS developers, to implement a > proven/stable SIP protocol stack. Nothing is perfect and that does not > mean that we did not spend a considerable amount of time documenting > bugs so that they could be addressed and it would work as it should > > We still use OpenSIPS for certain CSCF functionality (due to its speed > and flexibility since it is not a B2BUA). > > Based on Mathieu's response (and he is definitely someone that would > know), it looks like you should be able to easily append a X-AUTH-IP > header (via OpenSIPS) containing the IP address of the endpoint and call > it a day. > > -metik > > > Bill W wrote: >> Hey Metik, >> >> That's exactly what I'm trying to do... load balance across multiple FS >> boxes, and have any machine in the cluster be able to reach a device >> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >> >> True I could do the auth on the proxies as well, but then I'm setting up >> another authentication scheme in addition to what is on the FS boxes, >> and then integrating the databases so everything is consistent. >> >> I also have hosts that talk to the FS boxes directly, rather than >> through the proxy. So I can't get rid of auth_acl on FS either, even if >> I do implement it on the proxies. So my setup becomes much more >> complex and potentially brittle. >> >> And all we're really talking about for FreeSWITCH, conceptually >> speaking, is populating a variable with a different IP. We could even >> make it configurable, as to which IP is to be used for the auth-acl. >> >> What are you using for SBCs? (if you are allowed to divulge that) I'm >> currently using OpenSIPS for my proxy. >> >> Thanks, >> Bill >> >> Metik wrote: >> >>> Why not simply implement this feature in the PROXY itself? >>> >>> FS has a pretty comprehensive security feature set for endpoints that >>> directly register with it. >>> >>> Don't get me wrong, I do agree this is useful especially if you are >>> going to be using your proxies to load balance across multiple FS boxes >>> to create an ad-hoc cluster. I actually have session border controllers >>> that have this feature and use it quite often. >>> >>> -metik >>> >>> Bill W wrote: >>> >>>> Hey Metik, >>>> >>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>> >>>> I'll guess have to do that in the short term, but in my opinion, having >>>> auth-acl be able to work through a proxy is very important as it is a >>>> vital part of a comprehensive security feature set. And it would be >>>> much simpler to implement from an end-user perspective than the >>>> alternative of doing it in xml_curl. >>>> >>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>> What is the going rate for that kind of thing? >>>> >>>> Is anyone out there interested in coding this feature? Or chipping in >>>> for the bounty? >>>> >>>> >>>> Thanks, >>>> Bill >>>> >>>> >>>> Metik wrote: >>>> >>>> >>>>> This may be difficult considering that ACL needs to consider the >>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>> apply-proxy-acl. >>>>> >>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>> (and validating against appropriate variables available when using it to >>>>> authenticate the request). >>>>> >>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>> >>>>> -metik >>>>> >>>>> >>>>> Bill W wrote: >>>>> >>>>> >>>>>> Hey Brian, >>>>>> >>>>>> >>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>> through a proxy the way I want them to, even with setting >>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>> >>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>> >>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>> the proxy. >>>>>> >>>>>> >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> Brian West wrote: >>>>>> >>>>>> >>>>>> >>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Okay, I added: to my sofia >>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>> >>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>> I've got in >>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>> >>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>> >>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>> >>>>>>>> Thoughts? >>>>>>>> Thanks, >>>>>>>> Bill >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ron.freeswitch at mcleodnet.com Fri Dec 18 21:42:36 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Fri, 18 Dec 2009 21:42:36 -0800 Subject: [Freeswitch-users] Park with Pre Answer Message-ID: <6247DAB4ECC5499180AE946F843D5C09@fromage> Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091218/bb71c48e/attachment-0002.html From jason at jasonjgw.net Sat Dec 19 00:31:42 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 19 Dec 2009 19:31:42 +1100 Subject: [Freeswitch-users] RTP problems in recent revisions? Message-ID: <20091219083142.GA21558@jdc.jasonjgw.net> Revision 15904 is fine, but after upgrading to revision 16003 I get the following. 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). 2. A PCMU call to a SIP provider is fine for the first 20 to 30 seconds, then the audio breaks up completely. I have ZRTP compiled in, if that makes any difference. Obviously there's a regression somewhere. Let me know if I can provide further help. From mike at jerris.com Sat Dec 19 06:25:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Dec 2009 09:25:22 -0500 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <20091219083142.GA21558@jdc.jasonjgw.net> References: <20091219083142.GA21558@jdc.jasonjgw.net> Message-ID: <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Sat Dec 19 06:25:22 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 19 Dec 2009 09:25:22 -0500 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <20091219083142.GA21558@jdc.jasonjgw.net> References: <20091219083142.GA21558@jdc.jasonjgw.net> Message-ID: <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From anthony.minessale at gmail.com Sat Dec 19 07:19:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 09:19:54 -0600 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> References: <20091219083142.GA21558@jdc.jasonjgw.net> <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> Message-ID: <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> Also retest with no zrtp send a full console debug log with sip trace On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace. Mike On Dec 19, 2009, at 3:31 AM, Jason White wrote: > Revision 15904 is fine, but... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/5b11a82b/attachment-0002.html From anthony.minessale at gmail.com Sat Dec 19 08:27:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 10:27:21 -0600 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <6247DAB4ECC5499180AE946F843D5C09@fromage> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> Message-ID: <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> how are you parking it? do you have a debug log showing it happen? On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod wrote: > Is there any way to park a channel without causing pre-answer (resulting > is a SIP 183 Session Progress)? > > > > Thanks, > > Ron > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/3e841f1c/attachment-0002.html From freeswitch-users-list at metik.com Sat Dec 19 08:41:06 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sat, 19 Dec 2009 11:41:06 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2C6694.3060400@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> Message-ID: <4B2D0222.7060609@metik.com> Bill, I think you would add this to the user profile in the directory. The "brian.xml" example (located in ${confdir}/directory/) provided with the default/sample configuration files demonstrates how to to do this by introducing a "cidr" attribute to the the "user" element. Example: "http://wiki.freeswitch.org/wiki/Acl" contains some great info (including a relevant example). -metik Bill W. wrote: > Hey Metik, > > Thanks so much for your insights and your help. And yes, I was able to > append the X-AUTH-IP header with no problem. But that didn't solve the > issue. After some more research, it appears that the problem isn't with > auth-calls at all. > > I disabled all auth-call directives in every sip profile and the > registration through a proxy is still being rejected. > > I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the > ip variable against the auth_acl cidr. > > if (auth_acl) { > if (!switch_check_network_list_ip(ip, auth_acl)) { > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); > ret = AUTH_FORBIDDEN; > goto end; > } > > So I guess the question is, is it possible to control what gets put into > the ip variable? > > Thanks, > Bill > > > Metik wrote: > >> Honestly, several years ago I accomplished this by mod'ing SER (which >> became OpenSER which was then forked to OpenSIPS and Kamalio) and using >> one cluster of proxies for subscriber endpoints and another for >> infrastructure (so that I could keep RTP flows optimized yet support >> double NAT when required by an endpoint). Although the network looks >> different today. >> >> However, we were never quite happy about the lack of media failover >> (complicated NAT) and evaluated several commercial solutions until >> finding Covergence (which is now, for better or for worse since the jury >> is still out, owned by ACME Packet). At the time, they offered the best >> mix of security (their forte) yet scaled very well in comparison to >> their competitors that I had tested in our lab (ACME Packet, Kagoor, >> Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great >> decision, not unlike that of the FS developers, to implement a >> proven/stable SIP protocol stack. Nothing is perfect and that does not >> mean that we did not spend a considerable amount of time documenting >> bugs so that they could be addressed and it would work as it should >> >> We still use OpenSIPS for certain CSCF functionality (due to its speed >> and flexibility since it is not a B2BUA). >> >> Based on Mathieu's response (and he is definitely someone that would >> know), it looks like you should be able to easily append a X-AUTH-IP >> header (via OpenSIPS) containing the IP address of the endpoint and call >> it a day. >> >> -metik >> >> >> Bill W wrote: >> >>> Hey Metik, >>> >>> That's exactly what I'm trying to do... load balance across multiple FS >>> boxes, and have any machine in the cluster be able to reach a device >>> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >>> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >>> >>> True I could do the auth on the proxies as well, but then I'm setting up >>> another authentication scheme in addition to what is on the FS boxes, >>> and then integrating the databases so everything is consistent. >>> >>> I also have hosts that talk to the FS boxes directly, rather than >>> through the proxy. So I can't get rid of auth_acl on FS either, even if >>> I do implement it on the proxies. So my setup becomes much more >>> complex and potentially brittle. >>> >>> And all we're really talking about for FreeSWITCH, conceptually >>> speaking, is populating a variable with a different IP. We could even >>> make it configurable, as to which IP is to be used for the auth-acl. >>> >>> What are you using for SBCs? (if you are allowed to divulge that) I'm >>> currently using OpenSIPS for my proxy. >>> >>> Thanks, >>> Bill >>> >>> Metik wrote: >>> >>> >>>> Why not simply implement this feature in the PROXY itself? >>>> >>>> FS has a pretty comprehensive security feature set for endpoints that >>>> directly register with it. >>>> >>>> Don't get me wrong, I do agree this is useful especially if you are >>>> going to be using your proxies to load balance across multiple FS boxes >>>> to create an ad-hoc cluster. I actually have session border controllers >>>> that have this feature and use it quite often. >>>> >>>> -metik >>>> >>>> Bill W wrote: >>>> >>>> >>>>> Hey Metik, >>>>> >>>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>>> >>>>> I'll guess have to do that in the short term, but in my opinion, having >>>>> auth-acl be able to work through a proxy is very important as it is a >>>>> vital part of a comprehensive security feature set. And it would be >>>>> much simpler to implement from an end-user perspective than the >>>>> alternative of doing it in xml_curl. >>>>> >>>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>>> What is the going rate for that kind of thing? >>>>> >>>>> Is anyone out there interested in coding this feature? Or chipping in >>>>> for the bounty? >>>>> >>>>> >>>>> Thanks, >>>>> Bill >>>>> >>>>> >>>>> Metik wrote: >>>>> >>>>> >>>>> >>>>>> This may be difficult considering that ACL needs to consider the >>>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>>> apply-proxy-acl. >>>>>> >>>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>>> (and validating against appropriate variables available when using it to >>>>>> authenticate the request). >>>>>> >>>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>>> >>>>>> -metik >>>>>> >>>>>> >>>>>> Bill W wrote: >>>>>> >>>>>> >>>>>> >>>>>>> Hey Brian, >>>>>>> >>>>>>> >>>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>>> through a proxy the way I want them to, even with setting >>>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>>> >>>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>>> >>>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>>> the proxy. >>>>>>> >>>>>>> >>>>>>> Thanks, >>>>>>> Bill >>>>>>> >>>>>>> Brian West wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>>> >>>>>>>> /b >>>>>>>> >>>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> Okay, I added: to my sofia >>>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>>> >>>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>>> I've got in >>>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>>> >>>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>>> >>>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>>> >>>>>>>>> Thoughts? >>>>>>>>> Thanks, >>>>>>>>> Bill >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> ------------------------------------------------------------------------ >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Sat Dec 19 09:09:33 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 19 Dec 2009 18:09:33 +0100 Subject: [Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header In-Reply-To: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> References: <676BE5178B344371A73CDB4BE55216A8@greyhawk.tonecommander.com> Message-ID: <4B2D08CD.6060408@gmx.net> we do this based XML-Curl. Jerry Richards schrieb: > Is it possible to allow/deny REGISTER requests based on the User-Agent > header? I need to know/manage what devices are registering. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch-users-list at metik.com Sat Dec 19 09:47:23 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sat, 19 Dec 2009 12:47:23 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D0222.7060609@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> Message-ID: <4B2D11AB.3080906@metik.com> I noticed a typo in my post that may easily confuse someone... should be: -metik Metik wrote: > Bill, > > I think you would add this to the user profile in the directory. The > "brian.xml" example (located in ${confdir}/directory/) provided with the > default/sample configuration files demonstrates how to to do this by > introducing a "cidr" attribute to the the "user" element. > > Example: > > > > > > > > > > > > "http://wiki.freeswitch.org/wiki/Acl" contains some great info > (including a relevant example). > > -metik > > Bill W. wrote: > >> Hey Metik, >> >> Thanks so much for your insights and your help. And yes, I was able to >> append the X-AUTH-IP header with no problem. But that didn't solve the >> issue. After some more research, it appears that the problem isn't with >> auth-calls at all. >> >> I disabled all auth-call directives in every sip profile and the >> registration through a proxy is still being rejected. >> >> I looked in sofia_reg.c and if auth_acl is defined, sofia_reg checks the >> ip variable against the auth_acl cidr. >> >> if (auth_acl) { >> if (!switch_check_network_list_ip(ip, auth_acl)) { >> switch_log_printf(SWITCH_CHANNEL_LOG, >> SWITCH_LOG_WARNING, "IP %s Rejected by user acl %s\n", ip, auth_acl); >> ret = AUTH_FORBIDDEN; >> goto end; >> } >> >> So I guess the question is, is it possible to control what gets put into >> the ip variable? >> >> Thanks, >> Bill >> >> >> Metik wrote: >> >> >>> Honestly, several years ago I accomplished this by mod'ing SER (which >>> became OpenSER which was then forked to OpenSIPS and Kamalio) and using >>> one cluster of proxies for subscriber endpoints and another for >>> infrastructure (so that I could keep RTP flows optimized yet support >>> double NAT when required by an endpoint). Although the network looks >>> different today. >>> >>> However, we were never quite happy about the lack of media failover >>> (complicated NAT) and evaluated several commercial solutions until >>> finding Covergence (which is now, for better or for worse since the jury >>> is still out, owned by ACME Packet). At the time, they offered the best >>> mix of security (their forte) yet scaled very well in comparison to >>> their competitors that I had tested in our lab (ACME Packet, Kagoor, >>> Netrake, NexTone, Kagoor, and Jasomi). In fact, they made a great >>> decision, not unlike that of the FS developers, to implement a >>> proven/stable SIP protocol stack. Nothing is perfect and that does not >>> mean that we did not spend a considerable amount of time documenting >>> bugs so that they could be addressed and it would work as it should >>> >>> We still use OpenSIPS for certain CSCF functionality (due to its speed >>> and flexibility since it is not a B2BUA). >>> >>> Based on Mathieu's response (and he is definitely someone that would >>> know), it looks like you should be able to easily append a X-AUTH-IP >>> header (via OpenSIPS) containing the IP address of the endpoint and call >>> it a day. >>> >>> -metik >>> >>> >>> Bill W wrote: >>> >>> >>>> Hey Metik, >>>> >>>> That's exactly what I'm trying to do... load balance across multiple FS >>>> boxes, and have any machine in the cluster be able to reach a device >>>> behind a NAT firewall. Hence the need for the proxy. Also, I'm trying >>>> to keep the proxy relatively "dumb" and put all the logic in the FS boxes. >>>> >>>> True I could do the auth on the proxies as well, but then I'm setting up >>>> another authentication scheme in addition to what is on the FS boxes, >>>> and then integrating the databases so everything is consistent. >>>> >>>> I also have hosts that talk to the FS boxes directly, rather than >>>> through the proxy. So I can't get rid of auth_acl on FS either, even if >>>> I do implement it on the proxies. So my setup becomes much more >>>> complex and potentially brittle. >>>> >>>> And all we're really talking about for FreeSWITCH, conceptually >>>> speaking, is populating a variable with a different IP. We could even >>>> make it configurable, as to which IP is to be used for the auth-acl. >>>> >>>> What are you using for SBCs? (if you are allowed to divulge that) I'm >>>> currently using OpenSIPS for my proxy. >>>> >>>> Thanks, >>>> Bill >>>> >>>> Metik wrote: >>>> >>>> >>>> >>>>> Why not simply implement this feature in the PROXY itself? >>>>> >>>>> FS has a pretty comprehensive security feature set for endpoints that >>>>> directly register with it. >>>>> >>>>> Don't get me wrong, I do agree this is useful especially if you are >>>>> going to be using your proxies to load balance across multiple FS boxes >>>>> to create an ad-hoc cluster. I actually have session border controllers >>>>> that have this feature and use it quite often. >>>>> >>>>> -metik >>>>> >>>>> Bill W wrote: >>>>> >>>>> >>>>> >>>>>> Hey Metik, >>>>>> >>>>>> Thanks for the reply, and the pointers for doing it with xml_curl. >>>>>> >>>>>> I'll guess have to do that in the short term, but in my opinion, having >>>>>> auth-acl be able to work through a proxy is very important as it is a >>>>>> vital part of a comprehensive security feature set. And it would be >>>>>> much simpler to implement from an end-user perspective than the >>>>>> alternative of doing it in xml_curl. >>>>>> >>>>>> As a matter of fact, I'm considering offering a bounty for that feature. >>>>>> What is the going rate for that kind of thing? >>>>>> >>>>>> Is anyone out there interested in coding this feature? Or chipping in >>>>>> for the bounty? >>>>>> >>>>>> >>>>>> Thanks, >>>>>> Bill >>>>>> >>>>>> >>>>>> Metik wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> This may be difficult considering that ACL needs to consider the >>>>>>> original src IP/URI. To do that it, freeswitch would need to do so >>>>>>> using a header that retains that information (i.e. From, Via, Contact, >>>>>>> etc.). Which I do not believe is currently possible using auth-acl or >>>>>>> apply-proxy-acl. >>>>>>> >>>>>>> However, you should be able to emulate the behavior using mod_xml_curl >>>>>>> (and validating against appropriate variables available when using it to >>>>>>> authenticate the request). >>>>>>> >>>>>>> see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Authorization >>>>>>> >>>>>>> -metik >>>>>>> >>>>>>> >>>>>>> Bill W wrote: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> Hey Brian, >>>>>>>> >>>>>>>> >>>>>>>> I've been doing some testing and I am unable to get auth-calls to work >>>>>>>> through a proxy the way I want them to, even with setting >>>>>>>> apply-proxy-acl to either the endpoint IP or the proxy IP. >>>>>>>> >>>>>>>> I have a multi-tenant system with multiple domains with multiple users >>>>>>>> in each domain. And I want to restrict a user to an arbitrary CIDR and >>>>>>>> challenge them for a password. The arbitrary CIDR will vary from UA to >>>>>>>> UA, and is specified in the directory via the auth-acl parameter. >>>>>>>> >>>>>>>> TL,DR; I want to get auth-calls to use the IP of the UA endpoint, not of >>>>>>>> the proxy. >>>>>>>> >>>>>>>> >>>>>>>> Thanks, >>>>>>>> Bill >>>>>>>> >>>>>>>> Brian West wrote: >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> it needs to be an ACL from acl.conf or a ip/cidr >>>>>>>>> >>>>>>>>> /b >>>>>>>>> >>>>>>>>> On Dec 17, 2009, at 5:41 AM, Bill W wrote: >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>>> Okay, I added: to my sofia >>>>>>>>>> profile and restarted sofia, and still no joy. >>>>>>>>>> >>>>>>>>>> I'm on FreeSWITCH Version 1.0.trunk (15764) >>>>>>>>>> I've got in >>>>>>>>>> the directory, but I'm still being rejected by the acl: >>>>>>>>>> >>>>>>>>>> 2009-12-17 06:04:59.920517 [WARNING] sofia_reg.c:1928 IP 64.135.119.105 >>>>>>>>>> Rejected by user acl 190.218.103.12/32 >>>>>>>>>> >>>>>>>>>> Here's what I believe is the appropriate snippet of the debug output: >>>>>>>>>> http://pastebin.freeswitch.org/11531 >>>>>>>>>> >>>>>>>>>> Thoughts? >>>>>>>>>> Thanks, >>>>>>>>>> Bill >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> ------------------------------------------------------------------------ >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Sat Dec 19 10:29:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Dec 2009 12:29:10 -0600 Subject: [Freeswitch-users] RTP problems in recent revisions? In-Reply-To: <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> References: <20091219083142.GA21558@jdc.jasonjgw.net> <766E9E30-1504-4DB6-89E9-C5E8954B9F62@jerris.com> <191c3a030912190718x5d950e9eyb450847b9f7ee0ed@mail.gmail.com> <191c3a030912190719k5958e6d9g82af82533a6ec4fe@mail.gmail.com> Message-ID: <191c3a030912191029u5637df13v65509e42743af22d@mail.gmail.com> I tried a patch out of pure deduction and speculation from your post. Can you update and test it for me please? On Sat, Dec 19, 2009 at 9:19 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Also retest with no zrtp > send a full console debug log with sip trace > > On Dec 19, 2009 8:33 AM, "Michael Jerris" wrote: > > The best help to track this down is to try to identify the specific > svn revision that caused the issue and to supply a full freeswitch > debug with sip trace. > > Mike > > On Dec 19, 2009, at 3:31 AM, Jason White wrote: > > Revision 15904 is fine, but... > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/bfc974aa/attachment-0002.html From ron.freeswitch at mcleodnet.com Sat Dec 19 10:54:56 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 10:54:56 -0800 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> Message-ID: <347A9A7174A14557B5ADC2900B21A784@fromage> This is what I am doing . DIALPLAN NETWORK TRACE 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE sip:6042772011 at 192.168.100.132:5090, with session description 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy Authentication Required 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK sip:6042772011 at 192.168.100.132:5090 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE sip:6042772011 at 192.168.100.132:5090, with session description 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 Session Progress, with session description CONSOLE 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected by acl "domains". Falling back to Digest auth. 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected by acl "domains". Falling back to Digest auth. 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel sofia/internal/695 at 192.168.100.132:5060 [07a14700-eccf-11de-8080-6fed700309ce] 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: v=0 o=695 123456 654323 IN IP4 192.168.100.140 s=none c=IN IP4 192.168.100.140 t=0 0 m=audio 10900 RTP/AVP 0 8 18 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 (sofia/internal/695 at 192.168.100.132:5060) State NEW 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:115:32000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G7221:107:16000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/695 at 192.168.100.132:5060) State INIT 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> CS_ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/695 at 192.168.100.132:5060) State ROUTING 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing Phone 300->6042772011 in context mytest Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest->mytest] continue=false Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] destination_number(6042772011) =~ /^.*$/ break=on-false Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> CS_EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 Application park Requires media! pre_answering channel sofia/internal/695 at 192.168.100.132:5060 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending early media 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> 192.168.100.140 port 10900 codec: 0 ms: 20 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: v=0 o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 s=FreeSWITCH c=IN IP4 192.168.100.132 t=0 0 m=audio 25382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/695 at 192.168.100.132:5060! 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/695 at 192.168.100.132:5060 [BREAK] EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, December 19, 2009 8:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Park with Pre Answer how are you parking it? do you have a debug log showing it happen? On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod wrote: Is there any way to park a channel without causing pre-answer (resulting is a SIP 183 Session Progress)? Thanks, Ron _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/5ac5c7f1/attachment-0002.html From frank at carmickle.com Sat Dec 19 12:19:43 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sat, 19 Dec 2009 15:19:43 -0500 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <347A9A7174A14557B5ADC2900B21A784@fromage> References: <6247DAB4ECC5499180AE946F843D5C09@fromage> <191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com> <347A9A7174A14557B5ADC2900B21A784@fromage> Message-ID: <20091219201943.GG31924@base.carmickle.com> Good afternoon By default the internal profile is looking to have authed calls. If you want you can set an acl. Look at autoload_configs/acl.conf.xml. Also remember to set the context in the profile so that the dialplan for that context will be parsed. If you decide to register to it you can set the context in the directory entry for that user. Let us know how you do. --FC On Sat, Dec 19, Ron McLeod wrote: > This is what I am doing . > > > > DIALPLAN > > > > > > > > > > > > > > > > > > > > > > > > > > NETWORK TRACE > > 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > sip:6042772011 at 192.168.100.132:5090, with session description > > 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy > Authentication Required > > 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK > sip:6042772011 at 192.168.100.132:5090 > > 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > sip:6042772011 at 192.168.100.132:5090, with session description > > 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 Session > Progress, with session description > > > > > > CONSOLE > > 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/695 at 192.168.100.132:5060 > [07a14700-eccf-11de-8080-6fed700309ce] > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel > sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: > > v=0 > > o=695 123456 654323 IN IP4 192.168.100.140 > > s=none > > c=IN IP4 192.168.100.140 > > t=0 0 > > m=audio 10900 RTP/AVP 0 8 18 2 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:18 G729A/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:2 G726-32/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 > (sofia/internal/695 at 192.168.100.132:5060) State NEW > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:115:32000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G7221:107:16000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[G722:9:8000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > [PCMU:0:8000:20]/[PCMU:0:8000:20] > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec > sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload > to 101 > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/695 at 192.168.100.132:5060) State INIT > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 > sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> CS_ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 > sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 > sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING > > 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing Phone > 300->6042772011 in context mytest > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest->mytest] > continue=false > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] > destination_number(6042772011) =~ /^.*$/ break=on-false > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> > CS_EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 > sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 > sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 Application > park Requires media! pre_answering channel > sofia/internal/695 at 192.168.100.132:5060 > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 > sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] > > 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending early > media > > 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP > [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> > 192.168.100.140 port 10900 codec: 0 ms: 20 > > 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer [soft] > 160 bytes per 20ms > > 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: > > v=0 > > o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 > > s=FreeSWITCH > > c=IN IP4 192.168.100.132 > > t=0 0 > > m=audio 25382 RTP/AVP 0 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer > sofia/internal/695 at 192.168.100.132:5060! > > 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() > > 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel > sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] > > > > > > > > _____ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Saturday, December 19, 2009 8:27 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Park with Pre Answer > > > > how are you parking it? > do you have a debug log showing it happen? > > > > On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod > wrote: > > Is there any way to park a channel without causing pre-answer (resulting is > a SIP 183 Session Progress)? > > > > Thanks, > > Ron > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ron.freeswitch at mcleodnet.com Sat Dec 19 12:35:39 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 12:35:39 -0800 Subject: [Freeswitch-users] Park with Pre Answer In-Reply-To: <20091219201943.GG31924@base.carmickle.com> References: <6247DAB4ECC5499180AE946F843D5C09@fromage><191c3a030912190827h60dda9d6se500bccb1497d2fd@mail.gmail.com><347A9A7174A14557B5ADC2900B21A784@fromage> <20091219201943.GG31924@base.carmickle.com> Message-ID: <5872485EBB7443D1BFBC60D51CAF5578@fromage> My issue has nothing to do with registration or authentication. I am simply looking for a way to park a new call without having the call "pre-answered" (I don't want a SIP 183 sent back to the client). Ron > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Frank Carmickle > Sent: Saturday, December 19, 2009 12:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Park with Pre Answer > > Good afternoon > > By default the internal profile is looking to have authed calls. If you > want you can set an acl. Look at autoload_configs/acl.conf.xml. > > Also remember to set the context in the profile so that the dialplan for > that context will be parsed. If you decide to register to it you can set > the context in the directory entry for that user. > > Let us know how you do. > --FC > > > On Sat, Dec 19, Ron McLeod wrote: > > This is what I am doing . > > > > > > > > DIALPLAN > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > NETWORK TRACE > > > > 0.000000 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > > sip:6042772011 at 192.168.100.132:5090, with session description > > > > 0.000749 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > > > 0.053820 192.168.100.132 -> 192.168.100.140 SIP Status: 407 Proxy > > Authentication Required > > > > 0.185859 192.168.100.140 -> 192.168.100.132 SIP Request: ACK > > sip:6042772011 at 192.168.100.132:5090 > > > > 0.247509 192.168.100.140 -> 192.168.100.132 SIP/SDP Request: INVITE > > sip:6042772011 at 192.168.100.132:5090, with session description > > > > 0.248226 192.168.100.132 -> 192.168.100.140 SIP Status: 100 Trying > > > > 0.259591 192.168.100.132 -> 192.168.100.140 SIP/SDP Status: 183 > Session > > Progress, with session description > > > > > > > > > > > > CONSOLE > > > > 2009-12-19 10:47:59.556850 [DEBUG] sofia.c:4628 IP 192.168.100.140 > Rejected > > by acl "domains". Falling back to Digest auth. > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:4628 IP 192.168.100.140 > Rejected > > by acl "domains". Falling back to Digest auth. > > > > 2009-12-19 10:47:59.804984 [NOTICE] switch_channel.c:602 New Channel > > sofia/internal/695 at 192.168.100.132:5060 > > [07a14700-eccf-11de-8080-6fed700309ce] > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_NEW > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3289 Channel > > sofia/internal/695 at 192.168.100.132:5060 entering state [received][100] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3296 Remote SDP: > > > > v=0 > > > > o=695 123456 654323 IN IP4 192.168.100.140 > > > > s=none > > > > c=IN IP4 192.168.100.140 > > > > t=0 0 > > > > m=audio 10900 RTP/AVP 0 8 18 2 101 > > > > a=rtpmap:0 PCMU/8000 > > > > a=rtpmap:8 PCMA/8000 > > > > a=rtpmap:18 G729A/8000 > > > > a=fmtp:18 annexb=no > > > > a=rtpmap:2 G726-32/8000 > > > > a=rtpmap:101 telephone-event/8000 > > > > a=fmtp:101 0-15 > > > > a=ptime:20 > > > > > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_state_machine.c:404 > > (sofia/internal/695 at 192.168.100.132:5060) State NEW > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G7221:115:32000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G7221:107:16000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[G722:9:8000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3071 Audio Codec Compare > > [PCMU:0:8000:20]/[PCMU:0:8000:20] > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:2029 Set Codec > > sofia/internal/695 at 192.168.100.132:5060 PCMU/8000 20 ms 160 samples > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf > payload > > to 101 > > > > 2009-12-19 10:47:59.804984 [DEBUG] sofia.c:3455 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_NEW -> CS_INIT > > > > 2009-12-19 10:47:59.804984 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change CS_INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > > (sofia/internal/695 at 192.168.100.132:5060) State INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:83 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA INIT > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:111 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_INIT -> > CS_ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:481 > > (sofia/internal/695 at 192.168.100.132:5060) State INIT going to sleep > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change > CS_ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:130 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA ROUTING > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:78 > > sofia/internal/695 at 192.168.100.132:5060 Standard ROUTING > > > > 2009-12-19 10:47:59.812026 [INFO] mod_dialplan_xml.c:315 Processing > Phone > > 300->6042772011 in context mytest > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 parsing [mytest- > >mytest] > > continue=false > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Regex (PASS) [mytest] > > destination_number(6042772011) =~ /^.*$/ break=on-false > > > > Dialplan: sofia/internal/695 at 192.168.100.132:5060 Action park() > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:114 > > (sofia/internal/695 at 192.168.100.132:5060) State Change CS_ROUTING -> > > CS_EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:932 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:484 > > (sofia/internal/695 at 192.168.100.132:5060) State ROUTING going to sleep > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:398 > > (sofia/internal/695 at 192.168.100.132:5060) Running State Change > CS_EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:491 > > (sofia/internal/695 at 192.168.100.132:5060) State EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] mod_sofia.c:173 > > sofia/internal/695 at 192.168.100.132:5060 SOFIA EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_state_machine.c:151 > > sofia/internal/695 at 192.168.100.132:5060 Standard EXECUTE > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1390 > Application > > park Requires media! pre_answering channel > > sofia/internal/695 at 192.168.100.132:5060 > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_core_session.c:1392 > > sofia/internal/695 at 192.168.100.132:5060 receive message [PROGRESS] > > > > 2009-12-19 10:47:59.812026 [INFO] switch_core_session.c:1392 Sending > early > > media > > > > 2009-12-19 10:47:59.812026 [DEBUG] sofia_glue.c:2263 AUDIO RTP > > [sofia/internal/695 at 192.168.100.132:5060] 192.168.100.132 port 25382 -> > > 192.168.100.140 port 10900 codec: 0 ms: 20 > > > > 2009-12-19 10:47:59.812026 [DEBUG] switch_rtp.c:1138 Starting timer > [soft] > > 160 bytes per 20ms > > > > 2009-12-19 10:47:59.815157 [INFO] mod_sofia.c:1506 Ring SDP: > > > > v=0 > > > > o=FreeSWITCH 1261223097 1261223098 IN IP4 192.168.100.132 > > > > s=FreeSWITCH > > > > c=IN IP4 192.168.100.132 > > > > t=0 0 > > > > m=audio 25382 RTP/AVP 0 101 > > > > a=rtpmap:0 PCMU/8000 > > > > a=rtpmap:101 telephone-event/8000 > > > > a=fmtp:101 0-16 > > > > a=silenceSupp:off - - - - > > > > a=ptime:20 > > > > a=sendrecv > > > > > > > > 2009-12-19 10:47:59.815157 [NOTICE] mod_sofia.c:1509 Pre-Answer > > sofia/internal/695 at 192.168.100.132:5060! > > > > 2009-12-19 10:47:59.815157 [DEBUG] switch_core_session.c:630 Send signal > > sofia/internal/695 at 192.168.100.132:5060 [BREAK] > > > > EXECUTE sofia/internal/695 at 192.168.100.132:5060 park() > > > > 2009-12-19 10:47:59.818818 [DEBUG] sofia.c:3289 Channel > > sofia/internal/695 at 192.168.100.132:5060 entering state [early][183] > > > > > > > > > > > > > > > > _____ > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: Saturday, December 19, 2009 8:27 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Park with Pre Answer > > > > > > > > how are you parking it? > > do you have a debug log showing it happen? > > > > > > > > On Fri, Dec 18, 2009 at 11:42 PM, Ron McLeod > > > wrote: > > > > Is there any way to park a channel without causing pre-answer (resulting > is > > a SIP 183 Session Progress)? > > > > > > > > Thanks, > > > > Ron > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From magesh.freeswitch at gmail.com Fri Dec 18 22:18:28 2009 From: magesh.freeswitch at gmail.com (Magesh R) Date: Sat, 19 Dec 2009 11:48:28 +0530 Subject: [Freeswitch-users] Deleting event name in Filter Message-ID: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> Dear All, I have filter a Event like the following in Perl. $IVR->{_esl}->filter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); But I don't how to delete that filter by using filter method. Because 'filter' method accepts only two arguments. Could you any one tell me a way to do it? Thanks, Magesh. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091219/1d4582b5/attachment-0002.html From scottferri09 at gmail.com Sat Dec 19 10:30:57 2009 From: scottferri09 at gmail.com (Scott Fernandez) Date: Sun, 20 Dec 2009 00:00:57 +0530 Subject: [Freeswitch-users] Third Party device connectivity from Freeswitch Message-ID: Hi, Is there a way to have the Freeswitch to route the calls to physical device/phone rather than just routing the calls to soft phones like Xlite?. If available, What sort of settings that are required in Freeswitch to communicate with third party applications/hardwares (like PBX) so that the calls be switched to physical devices from Freeswitch? Can anyone help me in this regard?. Thanks, Scott. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/012b982d/attachment-0002.html From ron.freeswitch at mcleodnet.com Sat Dec 19 14:34:36 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 14:34:36 -0800 Subject: [Freeswitch-users] Deleting event name in Filter In-Reply-To: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> References: <369c72d80912182218o4b33e02aw398ecd155eb97bc0@mail.gmail.com> Message-ID: Would this work? $IVR->{_esl}->send("filter delete Event-Name CHANNEL_EXECUTE_COMPLETE"); Ron ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Magesh R Sent: Friday, December 18, 2009 10:18 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Deleting event name in Filter Dear All, ???? I have filter a Event like the following in Perl. ?$IVR->{_esl}->filter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); But I don't how to delete that filter by using filter method. Because 'filter' method accepts? only two arguments. Could you any one tell me a way to do it? Thanks, Magesh. -- This email was Anti Virus checked by Astaro Security Gateway. http://www.astaro.com From frank at carmickle.com Sat Dec 19 14:37:59 2009 From: frank at carmickle.com (Frank Carmickle) Date: Sat, 19 Dec 2009 17:37:59 -0500 Subject: [Freeswitch-users] Third Party device connectivity from Freeswitch In-Reply-To: References: Message-ID: <20091219223759.GI31924@base.carmickle.com> On Sun, Dec 20, Scott Fernandez wrote: > Hi, > > Is there a way to have the Freeswitch to route the calls to physical > device/phone rather than just routing the calls to soft phones like Xlite?. Very much so. It all depends on what you want to do. > If available, What sort of settings that are required in Freeswitch to > communicate with third party applications/hardwares (like PBX) so that the > calls be switched to physical devices from Freeswitch? Have a read through http://wiki.freeswitch.org/wiki/Interop_List HTH --FC From freeswitch at aastral.net Sat Dec 19 15:16:28 2009 From: freeswitch at aastral.net (Bill W.) Date: Sat, 19 Dec 2009 18:16:28 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D0222.7060609@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> Message-ID: <4B2D5ECC.4060209@aastral.net> Hey Metik, Yes. Well, actually, I can have the cidr in two places in the directory. >From what I understand the cidr= parmeter is used in conjunction with the apply-inbound-acl parameter in the sofia profile to just allow someone to make calls from a certain IP without authenticating. And from what I understand the auth-acl= parameter is used to restrict a user to a particular cidr, but the user has to authenticate as well. *The second feature is the one I want to use.* I want to force users to authenticate, but only allow that authentication from a particular cidr as an added measure against toll fraud. And this appears to be causing the issue. Because once I specify the auth-acl parameter in the directory, sofia-reg enforces that acl. And unfortunately it's using the IP of the proxy, not of the user-agent. I looked in sofia.c and found this comment: /* * if network_ip is a proxy allowed to send calls, check for auth * ip header and see if it matches against the inbound acl */ And this coincides with my testing. I have in my profile. I have my proxy sending the X-AUTH-IP header (verified with tcpdump). And yet the REGISTER is still being denied. So it appears that the apply-proxy-acl is set up to work with the apply-inbound-acl ( to allow users from an IP without authenticating) But that hasn't been carried over to sofia_reg.c, which appears to simply check the IP of who FreeSWITCH is talking to against the auth-acl cidr specified in the directory. (Line 1926) So I guess the question is, is my analysis correct? Thoughts anyone? Thanks, Bill Metik wrote: > Bill, > > I think you would add this to the user profile in the directory. The > "brian.xml" example (located in ${confdir}/directory/) provided with the > default/sample configuration files demonstrates how to to do this by > introducing a "cidr" attribute to the the "user" element. > > Example: > > > > > > > > > > > > "http://wiki.freeswitch.org/wiki/Acl" contains some great info > (including a relevant example). > > -metik > From ron.freeswitch at mcleodnet.com Sat Dec 19 17:29:31 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 17:29:31 -0800 Subject: [Freeswitch-users] Difference between ESL execute() and executeAsync() Message-ID: I don't notice any different in behavior between execute() and executeAsync(). I was expecting that executeAsync() would return right-away, and that execute() would only return after the specified application runs to completion (CHANNEL_EXECUTE_COMPLETE event). Running the sample app below, I see the "About to call execute(playback)" and "returned" displayed one right-after the other, even though the file being played takes about 4 minutes to play-out. Do I have this wrong, or is there something incorrect in my app? APP: #!/usr/bin/php events('plain', 'CHANNEL_STATE'); $eventSocket->filter('channel-state', 'CS_ROUTING'); // Wait for new call attempts while($eventSocket->connected()){ $event = $eventSocket->recvEvent(); $serializedBody = $event->serialize(); $listOfLines = toArrayOfLines($serializedBody); $nameValuePairs = toArrayOfNameValuePairs($listOfLines); $uuid = $nameValuePairs['Caller-Unique-ID']; printf("New call from uuid: $uuid\n"); // answer the caller and play announcement $eventSocket->execute('answer', Null ,$uuid); printf("About to call execute(playback)\n"); $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); printf("returned\n"); } ?> DIALPLAN: From ron.freeswitch at mcleodnet.com Sat Dec 19 19:16:03 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Sat, 19 Dec 2009 19:16:03 -0800 Subject: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() In-Reply-To: References: Message-ID: Here's the ES network trace: Content-Length: 1502 Content-Type: text/event-plain Event-Name: CHANNEL_STATE Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-19%2019%3A12%3A09 Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT Event-Date-Timestamp: 1261278729767397 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1024 Channel-State: CS_ROUTING Channel-State-Number: 2 Channel-Name: sofia/internal/699%40192.168.100.132 Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: ringing Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 699 Caller-Dialplan: XML Caller-Caller-ID-Name: Ron%20Soft%20Phone Caller-Caller-ID-Number: 699 Caller-Network-Addr: 192.168.100.3 Caller-Destination-Number: 444 Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Caller-Source: mod_sofia Caller-Context: mytest Caller-Channel-Name: sofia/internal/699%40192.168.100.132 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1261278729764077 Caller-Channel-Created-Time: 1261278729764077 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: answer execute-app-arg: Content-Type: command/reply Reply-Text: +OK sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: playback execute-app-arg: /tmp/ann.wav Content-Type: command/reply Reply-Text: +OK > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Saturday, December 19, 2009 5:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Difference between ESL execute() > andexecuteAsync() > > I don't notice any different in behavior between execute() and > executeAsync(). I was expecting that executeAsync() would return > right-away, and that execute() would only return after the specified > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > Running the sample app below, I see the "About to call execute(playback)" > and "returned" displayed one right-after the other, even though the file > being played takes about 4 minutes to play-out. > > Do I have this wrong, or is there something incorrect in my app? > > APP: > #!/usr/bin/php > require_once "ESL.php"; > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > $eventSocket->events('plain', 'CHANNEL_STATE'); > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > // Wait for new call attempts > while($eventSocket->connected()){ > $event = $eventSocket->recvEvent(); > $serializedBody = $event->serialize(); > $listOfLines = toArrayOfLines($serializedBody); > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > $uuid = $nameValuePairs['Caller-Unique-ID']; > printf("New call from uuid: $uuid\n"); > > // answer the caller and play announcement > $eventSocket->execute('answer', Null ,$uuid); > > printf("About to call execute(playback)\n"); > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > printf("returned\n"); > } > ?> > > > DIALPLAN: > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com From gabe at gundy.org Sat Dec 19 20:09:31 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Dec 2009 21:09:31 -0700 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Message-ID: <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> On Thu, Dec 17, 2009 at 6:14 AM, Hristo Benev wrote: > Just for your information there is a version of nmap for windows. So you can do the test from your desktop. Funny that you assume his desktop is running Windows (maybe it is). I would have guessed that the average person on this list doesn't run Windows on the desktop. But, what do I know? Gabe From jason at jasonjgw.net Sat Dec 19 20:18:31 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 20 Dec 2009 15:18:31 +1100 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> Message-ID: <20091220041831.GA9788@jdc.jasonjgw.net> Gabriel Gunderson wrote: > Funny that you assume his desktop is running Windows (maybe it is). I > would have guessed that the average person on this list doesn't run > Windows on the desktop. But, what do I know? Some of us on the list have never run Windows on anything. It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone via a USB head set, and also handling my Snom 320 SIP phone. From john_re at fastmail.us Sun Dec 20 03:13:17 2009 From: john_re at fastmail.us (john_re) Date: Sun, 20 Dec 2009 03:13:17 -0800 Subject: [Freeswitch-users] Dec 20 Global Freeswitch & All Free SW HW Culture meeting - BerkeleyTIP Message-ID: <1261307597.13829.1350992015@webmail.messagingengine.com> Hi FreeSwitchers, & Anthony Anthony - Thanks for letting me post the monthly announcement here. :) FSers: We are working toward moving to FS from Asterisk. We welcome you to join the BTIP Global VOIP bimonthly meetings, & if you like, help us get the FS sw running on our server. :) ===== A great December Solstice to you & yours. :) JOIN the Global All Free SW, HW, Culture meeting via VOIP Dec 20 Sunday, 12N-3PM (Pacific = UTC-8) = 3P-6P Eastern = 8P-11P UTC [Jan 2009 meetings: 2nd, 17th - mark your calendar] http://sites.google.com/site/berkeleytip/schedule == WATCH some VIDEOS: Mark Shuttleworth Interview - 10.04 Lucid Larynx Learning from Code History , Andreas Zeller Why does my program fail? Your version history might have the answer. Audio Hardware Enablement Session, UbuntuDevelopersSummit in Dallas Distributed Development, UDS in Dallas Splunk, Jeremy Thurgood CLUG Upstart, Stefano Rivera CLUG Interfacing with the real world, Mark Ter Morshuizen, Marc Welz CLUG Accelerating Graphics; Camp KDE 2009 http://sites.google.com/site/berkeleytip/talk-videos == Join the MAILING LIST & tell us which videos you will watch & why: http://groups.google.com/group/BerkTIPGlobal == JOIN the meeting via IRC & VOIP: Come discuss any & everything, & work on your individual or group projects. HOT TOPICS: Ub or KUb 9.10?, Ubuntu 10.04 plans, Android, Python3000 in 2010? Start on the #berkeleytip irc.freenode.net channel, & we'll help you get your VOIP system up & working. For VOIP SW, & connection info, see: http://sites.google.com/site/berkeleytip/remote-attendance Berkeley meeting LOCATION: Watch the website & mail list for latest details, perhaps at the Berkeley Public Library, or a cafe, due to Free Speech Cafe closed for winter break. http://sites.google.com/site/berkeleytip/ == OPPORTUNITIES to VOLUNTEER or learn new JOB SKILLS for 2010: Help set up our: Mailing list, FreeSwitch VOIP server, website http://sites.google.com/site/berkeleytip/opportunities Inquire & discuss at the meeting. == For Forwarding - You are invited to forward this announcement wherever it would be appreciated. From mcampbellsmith at gmail.com Sun Dec 20 03:58:02 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 20 Dec 2009 22:58:02 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: Message-ID: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2001 at 192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks From yehavi.bourvine at gmail.com Sun Dec 20 06:26:07 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 20 Dec 2009 16:26:07 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: Peer did not provide X.509 Certificate I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. Thanks! __Yehavi: 2009/12/17 Yehavi Bourvine > I am trying Audiocodes and Vegastream ATAs, and work with either the > manufacturer or the local representative here. > On SNOM I managed to make it work, and will try Polycom soon (once I manage > to grab one unit from our users...). > > Thanks, __yehavi: > > 2009/12/17 Brian West > >> Also what device are you using? I haven't tested with many so far... >> Polycom, Snom and a few others do TLS (see interop page on wiki) others do >> it wrong. >> >> /b >> >> On Dec 17, 2009, at 10:04 AM, Kristian Kielhofner wrote: >> >> You could try ssldump: >> >> http://www.rtfm.com/ssldump/ >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/1295a7eb/attachment-0002.html From JCasale at activenetwerx.com Sun Dec 20 07:58:12 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Sun, 20 Dec 2009 15:58:12 +0000 Subject: [Freeswitch-users] fs_cli connection error Message-ID: Trying to setup a new config in the pfSense 1.2.3 final package and when I try to connect to the console I get an auth error? # ./fs_cli -H 10.0.0.1 [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Authentication Error] I tried to search for docs to indicate where one might set the password for this (it never used to have one) but I could only see docs suggesting to provide one, not set one. There is no .fs_cli_conf anywhere. Socketstat shows it listening on 8021... Thanks! jlc From a.afzali2003 at gmail.com Sun Dec 20 08:08:27 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 20 Dec 2009 19:38:27 +0330 Subject: [Freeswitch-users] Interfacing to RabbitMQ Message-ID: Hi, I'll appreciate if someone who has a practice in interfacing FreeSWITCH to RabbitMQ or suggestions could share it to me. Regards, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/2b55e061/attachment-0002.html From msc at freeswitch.org Sun Dec 20 11:41:01 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 20 Dec 2009 11:41:01 -0800 Subject: [Freeswitch-users] fs_cli connection error In-Reply-To: References: Message-ID: The password is set in conf/autoload_configs/event_socket.conf.xml -MC Sent from my iPhone On Dec 20, 2009, at 7:58 AM, "Joseph L. Casale" wrote: > Trying to setup a new config in the pfSense 1.2.3 final package and > when > I try to connect to the console I get an auth error? > > # ./fs_cli -H 10.0.0.1 > [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting > [Authentication Error] > > I tried to search for docs to indicate where one might set the > password for > this (it never used to have one) but I could only see docs > suggesting to provide > one, not set one. > > There is no .fs_cli_conf anywhere. Socketstat shows it listening on > 8021... > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From andrew at hijacked.us Sun Dec 20 11:46:52 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 20 Dec 2009 14:46:52 -0500 Subject: [Freeswitch-users] Interfacing to RabbitMQ In-Reply-To: References: Message-ID: <20091220194651.GB1956@hijacked.us> On Sun, Dec 20, 2009 at 07:38:27PM +0330, afshin afzali wrote: > Hi, > > I'll appreciate if someone who has a practice in interfacing FreeSWITCH to > RabbitMQ or suggestions could share it to me. > You could try to use mod_erlang_event and the erlang rabbitmq client (in native message passing mode). I've never worked with rabbitMQ however, I just know a little about it. Andrew From a.alalousi at gmail.com Sun Dec 20 12:19:27 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Sun, 20 Dec 2009 20:19:27 +0000 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: References: Message-ID: People, Please do excuse me if this is a FAQ. I've so far not worked out a way to implement IP authentication effectively. I have a number of gateways/end points/...etc. hitting the switch without registration and originating calls to a number of upstreams I have configured. So far, everything is smooth and FREESwitch is legendary in every way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon machine running CentOS and CPU usage is barely a blip. My problem is authentication. I need a method to authenticate end points based only on their IP addresses. I am currently filtering access through the firewalls, but I would really like to delegate this task to FS in prep for an SBC setup I'm working on. If I remove the firewall filters, then anyone is able to get in. I could't workout precisely how ACLs work in FS from the WiKi and the documentation, and I haven't been able to make sense of how Digest functions either. Can anyone shed some light on those two areas ? I would like to really get to the bottom of this and update the WiKi pages with the working setup once done. Regards, Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/688baec0/attachment-0002.html From mrene_lists at avgs.ca Sun Dec 20 12:26:08 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 20 Dec 2009 15:26:08 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: References: Message-ID: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> Check out: http://wiki.freeswitch.org/wiki/ACL#Users It'll automatically add users with a cidr= attribute to the ACL list. This way you can set channel variables in the users and use them through your dialplan, all authenticated by ip address. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 20-Dec-09, at 3:19 PM, Ahmed Naji wrote: > People, > > Please do excuse me if this is a FAQ. > > I've so far not worked out a way to implement IP authentication > effectively. I have a number of gateways/end points/...etc. hitting > the switch without registration and originating calls to a number of > upstreams I have configured. > > So far, everything is smooth and FREESwitch is legendary in every > way. Over 10,000 simultaneous sessions at 120cps, on a dual-Xeon > machine running CentOS and CPU usage is barely a blip. > > My problem is authentication. I need a method to authenticate end > points based only on their IP addresses. I am currently filtering > access through the firewalls, but I would really like to delegate > this task to FS in prep for an SBC setup I'm working on. If I remove > the firewall filters, then anyone is able to get in. > > I could't workout precisely how ACLs work in FS from the WiKi and > the documentation, and I haven't been able to make sense of how > Digest functions either. > > Can anyone shed some light on those two areas ? I would like to > really get to the bottom of this and update the WiKi pages with the > working setup once done. > > Regards, > > Ahmed. > > > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/24e95788/attachment-0002.html From freeswitch-users-list at metik.com Sun Dec 20 13:58:52 2009 From: freeswitch-users-list at metik.com (Metik) Date: Sun, 20 Dec 2009 16:58:52 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2D5ECC.4060209@aastral.net> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> <4B2D5ECC.4060209@aastral.net> Message-ID: <4B2E9E1C.8090909@metik.com> Then it would appear that my original suggestion to use mod_xml_curl would be best for now and you may need to offer a bounty for this feature as others have suggested. Based on the sofia related snippets presented--I would assume it would be trivial to implement since most of the functionality is already there it just needs to be enhanced for your purpose. It would also be extremely easy to do this in OpenSIPS as well (using blacklists or avpops). Just so that I understand your dilemna, you want to reject an incoming REGISTER associated with a specific user unless it comes from a fixed location and if it does, you want to simply challenge it as usual to prevent toll fraud? I have found that its best to mitigate an attack at ingress before it even makes it to critical infrastructure (media gateways, application/media servers, etc.). -metik Bill W. wrote: > Hey Metik, > > Yes. Well, actually, I can have the cidr in two places in the directory. > > > > > > >From what I understand the cidr= parmeter is used in conjunction with > the apply-inbound-acl parameter in the sofia profile to just allow > someone to make calls from a certain IP without authenticating. > > And from what I understand the auth-acl= parameter is used to restrict a > user to a particular cidr, but the user has to authenticate as well. > > *The second feature is the one I want to use.* I want to force users to > authenticate, but only allow that authentication from a particular cidr > as an added measure against toll fraud. > > And this appears to be causing the issue. Because once I specify the > auth-acl parameter in the directory, sofia-reg enforces that acl. And > unfortunately it's using the IP of the proxy, not of the user-agent. > > I looked in sofia.c and found this comment: > /* > * if network_ip is a proxy allowed to send calls, check for auth > * ip header and see if it matches against the inbound acl > */ > > And this coincides with my testing. > I have in my > profile. I have my proxy sending the X-AUTH-IP header (verified with > tcpdump). And yet the REGISTER is still being denied. > > So it appears that the apply-proxy-acl is set up to work with the > apply-inbound-acl ( to allow users from an IP without authenticating) > > But that hasn't been carried over to sofia_reg.c, which appears to > simply check the IP of who FreeSWITCH is talking to against the auth-acl > cidr specified in the directory. (Line 1926) > > So I guess the question is, is my analysis correct? > > Thoughts anyone? > > Thanks, > Bill > > > > > > > Metik wrote: > >> Bill, >> >> I think you would add this to the user profile in the directory. The >> "brian.xml" example (located in ${confdir}/directory/) provided with the >> default/sample configuration files demonstrates how to to do this by >> introducing a "cidr" attribute to the the "user" element. >> >> Example: >> >> >> >> >> >> >> >> >> >> >> >> "http://wiki.freeswitch.org/wiki/Acl" contains some great info >> (including a relevant example). >> >> -metik >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sun Dec 20 14:28:06 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:28:06 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Message-ID: You'll need to fix your device to know its IP and it should stop doing that. /b On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote: > Hi! > > I'm sure this is a NAT issue, but I'm not sure what options to use. > > I have a Linksys SPA3102, NAT'd on the internet (remotely) and > connected to my FS on the otherside of the world, which is also > natted. A PAP2T is connected on the same subnet as the FS. The 3102 > registers successfully and a call can be set up from the PAP2 to the > 3102. > > However, after FS receives the Remote SDP the audio stops (ring tone > stops in my case) > > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel > sofia/internal/sip:2001 at 192.168.1.3:56885 entering state > [completing][200] > 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: > v=0 > o=- 18490612 18490612 IN IP4 192.168.1.3 > s=- > c=IN IP4 192.168.1.3 > t=0 0 > m=audio 16432 RTP/AVP 2 100 101 > a=rtpmap:2 G726-32/8000 > a=rtpmap:100 NSE/8000 > a=fmtp:100 192-193 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > I notice that the ip address in the o and c fields indicate a local IP > address. Should this IP address be an external IP address of the 3102 > instead? > > Thanks From freeswitch at skillsaw.com Sun Dec 20 12:53:08 2009 From: freeswitch at skillsaw.com (Gad Bentolila) Date: Sun, 20 Dec 2009 15:53:08 -0500 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> Message-ID: <4B2E8EB4.2040105@skillsaw.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/3a32b142/attachment-0002.html From brian at freeswitch.org Sun Dec 20 14:36:14 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:36:14 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <4B2E8EB4.2040105@skillsaw.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> Message-ID: <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: > DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt. Welcome to the community. > I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). > > Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). Your endpoint need only insert rport and FreeSWITCH will do the right thing. > 1) Setup stun on your remote endpoint (spa3102 in your case) > 2) Add to the directory xml file that describes your spa3102 endpoint The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar. > Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? > > I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. > > Anyway, hope this helps you with your SPA3102. Bottom line is enable rport and use stun on the SPA and it'll just work. /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/f773fada/attachment-0002.html From brian at freeswitch.org Sun Dec 20 14:45:23 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:45:23 -0600 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> Message-ID: <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> You have to watch it with TLS. Make sure your distro didn't mess up your SSL libs due to the recent vulnerability found. I havn't tested with my polycom in a few weeks but it was working on my Polycom after I uploaded the ca cert and marked it as trusted/used on the phone. /b On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: > I am trying now to set a Polycom to work with FreeSwitch and TLS. I have a Polycom-501 which does not have an internal certificate, thus only one-way certificate validation is needed. I've downloaded the root certificate to he Polyciom, and Freeswitch gives me the following error: > > Peer did not provide X.509 Certificate > I understand that it tries to do mutual authentication which is not possible in this case. How can I tell FreeSwitch to ignore the client's certificate? > > BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and Yealink. > > Thanks! __Yehavi: From brian at freeswitch.org Sun Dec 20 14:46:47 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Dec 2009 16:46:47 -0600 Subject: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports? In-Reply-To: <20091220041831.GA9788@jdc.jasonjgw.net> References: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <903da5680912192009h751df5c8h94771850374add91@mail.gmail.com> <20091220041831.GA9788@jdc.jasonjgw.net> Message-ID: The funny part is... it won't matter. Their are times when people post questions or issues and its well into debugging the issue before we realize "oh, you're on windows?". For the most part the windows installer is one of the most popular files on our website. /b On Dec 19, 2009, at 10:18 PM, Jason White wrote: > Gabriel Gunderson wrote: > >> Funny that you assume his desktop is running Windows (maybe it is). I >> would have guessed that the average person on this list doesn't run >> Windows on the desktop. But, what do I know? > > Some of us on the list have never run Windows on anything. > > It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone > via a USB head set, and also handling my Snom 320 SIP phone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/21b28f46/attachment-0002.html From mcampbellsmith at gmail.com Sun Dec 20 15:54:43 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Mon, 21 Dec 2009 10:54:43 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> Message-ID: <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> Thanks Brian and Gad, I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102. These are the options that I have set (according to the 3102 manual). ? Handle VIA received: yes ? Handle VIA rport: yes ? Insert VIA received: yes ? Insert VIA rport: yes ? Substitute VIA Addr: yes ? Send Resp To Src Port: yes ? STUN Enable: Choose yes. ? STUN Server: stun.freeswitch.org I assume that is all is needed? On Mon, Dec 21, 2009 at 9:36 AM, Brian West wrote: > > On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote: > > DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a > grain of salt. > > Welcome to the community. > > I have a similar setup (and problem) - the wiki documentation refers to it > as "double nat". Like you, my FS and client are behind different NATs and I > can register my remote endpoint and make calls (in my case, to the the FS > demo ivr at 5000). > > Since your external endpoint (spa3102) is registering, you've likely setup > your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). > > Your endpoint need only insert rport and FreeSWITCH will do the right thing. > > > 1) Setup stun on your remote endpoint (spa3102 in your case) > 2) Add value="NDLB-connectile-dysfunction"/> to the directory xml file that > describes your spa3102 endpoint > > The device supports STUN also its highly recommended your device know how to > overcome its own NAT. ?I personally do not believe its the registrars place > to overcome an endpoints?nat... puts undue burden on the registar. > > Option 1 worked for me right away (eyebeam in my case) and, as expected, the > remote sdp had the correct (remote) IP address, since the endpoint is using > stun to correctly identify its IP address to FS. However, option 2 has not > made a difference (for me). Is it just me or is it strange that SIP works > without stun, but RTP doesn't? > > I guess I've been spoiled by the way Asterisk handles NAT and was hopeful > that?NDLB-connectile-dysfunction would behave similarly, so I wouldn't have > to tell users to setup stun on their clients.?Maybe a FS user with some > experience with this type of NAT setup and these settings can help. I'd be > interested in knowing how to correctly setup remote NATted endpoints without > stun - or, at least, hear from someone that this setting works for them > without stun. > > Anyway, hope this helps you with your SPA3102. > > Bottom line is enable rport and use stun on the SPA and it'll just work. > /b > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From darklion11 at yahoo.com Sun Dec 20 18:39:54 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 18:39:54 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan Message-ID: <26868725.post@talk.nabble.com> Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26868725.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Sun Dec 20 19:47:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Sun, 20 Dec 2009 19:47:25 -0800 Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <87f2f3b90912201947i46483334kd69938de0446c456@mail.gmail.com> On Sun, Dec 20, 2009 at 6:39 PM, Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be > put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > This functionality already exists in the default dialplan and sample directory entries, assuming that you are using authorization. First off, look in 1000.xml (or any of the other sample user files) for this variable declaration: For any user whom you wish to restrict to local or domestic calling only just remove the 'international' from the list: Now when that user registers and makes calls he/she won't have 'international' in the ${toll_allow} channel variable. Something like this in your dialplan could handle both cases: Now that I've typed all that, I should go back and ask: are you using digest authorization? Or are you using an ACL to let your callers in? Anyway, hopefully the above example will give you some ideas. -MC > Thanks, > Edmar > -- > View this message in context: > http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26868725.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/c9c70b29/attachment-0002.html From darklion11 at yahoo.com Sun Dec 20 21:17:55 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 21:17:55 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan Message-ID: <26869283.post@talk.nabble.com> Hi Sir, How can I allow international calling in the dialing plan but for select accounts only? For example i want to restrict 8555555 to call this ip address 182.138.252.12 using the default configuration.. Does this command should be put in the default.xml or in the default folder and the filename is 00_restict.xml? When i tried this command both of them nothing happen 8555555 can call 182.138.252.12 i want it to restrict this account for not calling 182.138.252.12.. Please help.. Thanks, Edmar -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26869283p26869283.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Dec 20 21:55:36 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 21:55:36 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <26870199.post@talk.nabble.com> Not actually for now... I commented first the ACL restrictions... Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > > Thanks, > Edmar > -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26870199.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From senakahks at gmail.com Sun Dec 20 22:08:34 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Sun, 20 Dec 2009 22:08:34 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: Message-ID: <1261375714756-4197038.post@n2.nabble.com> Hi, I got it working. Can somebody explain me this error: 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: [Missing local host]. Also I am confused about heartbeat rate. Is enable_heartbeat_events=5 setting the heartbeat to 5? Thank you in advance, Amarakeerthi S wrote: > > Dear Sir, > > I have successfully installed freeSWITCH and it works fine in passthrough > mode. I installed nibblebill and it deduct money from the accounts > database > and it works fine. but I have two problems. > > 1. Calls can be initiated even though there is a minus value in accounts > database > > 2. Calls doesn't hangup when it goes to minus values. > > Any answers are greatly appreciated. > > This is my dialplan: > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Sun Dec 20 22:14:50 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 21 Dec 2009 16:14:50 +1000 Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: <1261375714756-4197038.post@n2.nabble.com> References: <1261375714756-4197038.post@n2.nabble.com> Message-ID: what did you have to change, to get this working ? Jay On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S wrote: > > Hi, > > I got it working. > > Can somebody explain me this error: > > 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: > [Missing local host]. Also I am confused about heartbeat rate. Is > enable_heartbeat_events=5 setting the heartbeat to 5? > > > Thank you in advance, > > > > > Amarakeerthi S wrote: > > > > Dear Sir, > > > > I have successfully installed freeSWITCH and it works fine in passthrough > > mode. I installed nibblebill and it deduct money from the accounts > > database > > and it works fine. but I have two problems. > > > > 1. Calls can be initiated even though there is a minus value in accounts > > database > > > > 2. Calls doesn't hangup when it goes to minus values. > > > > Any answers are greatly appreciated. > > > > This is my dialplan: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> > > > > > > > > > > > > This is the configuration file; > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/0d3b9dda/attachment-0002.html From darklion11 at yahoo.com Sun Dec 20 22:31:13 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 20 Dec 2009 22:31:13 -0800 (PST) Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26868725.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> Message-ID: <26870380.post@talk.nabble.com> Where should I write this line on the default.xml or in the default category? Edmar Cruz wrote: > > Hi Sir, > > How can I allow international calling in the dialing plan but for > select accounts only? > > For example i want to restrict 8555555 to call this ip address > 182.138.252.12 using the default configuration.. Does this command should > be put in the default.xml or in the default folder and the filename is > 00_restict.xml? > > > > > data="effective_caller_name=${effective_caller_id_name}"/> > data="effective_caller_number=${effective_caller_id_number}"/> > > > > > > > When i tried this command both of them nothing happen 8555555 can call > 182.138.252.12 i want it to restrict this account for not calling > 182.138.252.12.. > > Please help.. > > Thanks, > Edmar > -- View this message in context: http://old.nabble.com/Setting-Restrictions-on-Default-Dialplan-tp26868725p26870380.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From talk2ram at gmail.com Sun Dec 20 23:03:46 2009 From: talk2ram at gmail.com (ram) Date: Sun, 20 Dec 2009 23:03:46 -0800 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091219031649.GA1956@hijacked.us> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> Message-ID: Hi its good to hear any compare document between Vicidial and this project Ram On Fri, Dec 18, 2009 at 7:16 PM, Andrew Thompson wrote: > I've been asked to provide some screenshots, so here's some of the > agent/supervisor interface: > > http://eagle.bsd.st/~andrew/cpxshots/ > > Hopefully the image names are self-explanatory. In the ringing picture, > that URL pop is a configurable URL that can be used to integrate with a > CRM, in my case our own CRM - spicecsm. The URL supports interpolation > for variables like callerid, clientid, call type, etc. > > The supervisor view is a little hard to describe via static images, but > you're able to drag and drop agents into another profile (empty profiles > are hidden when not dragging an agent), drag agents onto an agent to > send them the call, and there's also various right click menus > available. > > Oh, and I forgot to mention this before; this system is in 'live > testing' and the goal is to do a final deployment sometime in January. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091220/ecc6716d/attachment-0002.html From ron.freeswitch at mcleodnet.com Mon Dec 21 00:48:35 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Mon, 21 Dec 2009 00:48:35 -0800 Subject: [Freeswitch-users] How can I detect an execute failure using ESL? Message-ID: <0A22A70A58F642D0A3B778A11C17A67C@fromage> When I try and perform an operation on a channel which has gone, an error is returned. How can I detect this using the ESL? execute() and sendRecv() always return 0 (zero) regardless of whether the command returns +OK or -ERR. sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533 call-command: execute execute-app-name: hangup execute-app-arg: UNALLOCATED_NUMBER Content-Type: command/reply Reply-Text: -ERR invalid session id [5d09753c-ede7-11de-85c6-27ab474dd533] Thanks, Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/8a916666/attachment-0002.html From Prometheus001 at gmx.net Mon Dec 21 04:02:51 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 21 Dec 2009 13:02:51 +0100 Subject: [Freeswitch-users] No Ringing tone when call is forwarded to PSTN In-Reply-To: References: <4B28D6FD.6010702@gmx.net> <4B2B52D7.9030505@gmx.net> Message-ID: <4B2F63EB.9030608@gmx.net> I just crosschecked the dialplan which is used. We do not anwer the call, we bridge it directly to a PSTN destination. However the Ringing event is not passed to PSTN(A): > PSTN(A)====INVITE===>FS > PSTN(A)<===TRYING===>FS > FS===INVITE==>PSTN(B) > FS<==TRYING===PSTN(B) > FS<==RINGING==PSTN(B) > PSTN(A)<==PROGRESS===FS > FS<===OK======PSTN(B) > FS====ACK====>PSTN(B) > PSTN(A)<===OK========FS > PSTN(A)====ACK======>FS But then I stumbled over the following SOFIA LOOPBACK entry in the logs: 2009-12-21 12:47:00.404145 [DEBUG] switch_core_state_machine.c:351 (sofia/external/06322xxxxxxxxxx at 10.11.12.15) State XCHANGE_MEDIA 2009-12-21 12:47:00.404145 [DEBUG] mod_sofia.c:469 SOFIA LOOPBACK 2009-12-21 12:47:00.404145 [DEBUG] sofia.c:3669 Channel sofia/external/0171xxxxxxx at 10.11.12.15:5060 skipping state [early][183] So I modified the dialplan to temporarily use another Patton GW for outgoing calls, et voil?, I receive a ringing tone at PSTN(A). So I think this is because Freeswitch thinks this is a loopback, because incoming and outgoing gateway is the same. But I due to other restrictions we need the call to pass through the same Patton Gateway to PSTN(B) as we received it from PSTN(A). Is there a chance to tell Freeswitch to not consider this call as a loopback scenario? Best regards Peter Brian West schrieb: > That depends if the call is answered and then you transfer it, you will HAVE to set the transfer_ringback variable you can't send a 180 to the thing or a progress and make it generate the ringback. You MUST do it yourself. > > You also fail to mention if the progress is a 180 or a 183 with sdp and media... or even better a 180 with sdp and media (silly sip people what were you thinking) either way... set the transfer_ringback variable. > > /b > > On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote: > > >> Should I open a JIRA for this? >> >> Best regards >> Peter >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Mon Dec 21 07:03:58 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 09:03:58 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> Message-ID: <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> Can you get me siptraces please. /b On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote: > Thanks Brian and Gad, > > I have stun set and if I do a 'sofia status profile internal', I see > the external IP address of the 3102 ATA, so I assume that stun is > working correctly on the SPA3102. > > These are the options that I have set (according to the 3102 manual). > > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server: stun.freeswitch.org > > I assume that is all is needed? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/833b0c89/attachment-0002.html From senakahks at gmail.com Mon Dec 21 07:36:31 2009 From: senakahks at gmail.com (Amarakeerthi S) Date: Mon, 21 Dec 2009 07:36:31 -0800 (PST) Subject: [Freeswitch-users] Mod nibblebill deduct money but no hangup at zero and can call without money in database In-Reply-To: References: <1261375714756-4197038.post@n2.nabble.com> Message-ID: I have changed the changed the dialplan little bit (thank to the people at IRC #freeswitch) as follows. Here I don't understand enable_heartbeat_events=5. It may be the heartbeat rate. Also I am getting this error in FS cli 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS ERROR: [Missing local host] On Tue, Dec 22, 2009 at 12:12 AM, jay binks [via freeswitch-users] wrote: > what did you have to change, to get this working ? > Jay > > On Mon, Dec 21, 2009 at 4:08 PM, Amarakeerthi S <[hidden email]> wrote: >> >> Hi, >> >> I got it working. >> >> Can somebody explain me this error: >> >> 2009-12-21 00:37:14.886242 [ERR] sofia_glue.c:2710 AUDIO RTP REPORTS >> ERROR: >> [Missing local host]. Also I am confused about heartbeat rate. Is >> enable_heartbeat_events=5 ?setting the heartbeat to 5? >> >> >> Thank you in advance, >> >> >> >> >> Amarakeerthi S wrote: >> > >> > Dear Sir, >> > >> > I have successfully installed freeSWITCH and it works fine in >> > passthrough >> > mode. I installed nibblebill and it deduct money from the accounts >> > database >> > and it works fine. but I have two problems. >> > >> > 1. Calls can be initiated even though there is a minus value in accounts >> > database >> > >> > 2. Calls doesn't hangup when it goes to minus values. >> > >> > Any answers are greatly appreciated. >> > >> > This is my dialplan: >> > >> > >> > >> > >> > ? >> > ? ? >> > ? ? >> > ? >> > >> > ? >> > >> > >> > >> > >> > >> > > > data="{absolute_codec_string=g729}sofia/gateway/OMEGA/5544$1"/> >> > >> > >> > >> > >> > >> > This is the configuration file; >> > >> > >> > ? >> > ? ? >> > >> > ? ? >> > >> > >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > >> > ? ? >> > >> > >> > >> > ? >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > [hidden email] >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4197038.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely > > Jay > > _______________________________________________ > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > View message @ > http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4198003.html > To unsubscribe from Re: Mod nibblebill deduct money but no hangup at zero > and can call without money in database, click here. > -- View this message in context: http://n2.nabble.com/Mod-nibblebill-deduct-money-but-no-hangup-at-zero-and-can-call-without-money-in-database-tp4174333p4198998.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/b6828d92/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 21 07:42:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 09:42:30 -0600 Subject: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() In-Reply-To: References: Message-ID: <191c3a030912210742y108f764et83581b50bd2bd60b@mail.gmail.com> if you run the socket in async mode, every call to execute is async if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod wrote: > Here's the ES network trace: > > Content-Length: 1502 > Content-Type: text/event-plain > Event-Name: CHANNEL_STATE > Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-19%2019%3A12%3A09 > Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT > Event-Date-Timestamp: 1261278729767397 > Event-Calling-File: switch_channel.c > Event-Calling-Function: switch_channel_perform_set_running_state > Event-Calling-Line-Number: 1024 > Channel-State: CS_ROUTING > Channel-State-Number: 2 > Channel-Name: sofia/internal/699%40192.168.100.132 > Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f > Call-Direction: inbound > Presence-Call-Direction: inbound > Answer-State: ringing > Channel-Read-Codec-Name: PCMU > Channel-Read-Codec-Rate: 8000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Caller-Username: 699 > Caller-Dialplan: XML > Caller-Caller-ID-Name: Ron%20Soft%20Phone > Caller-Caller-ID-Number: 699 > Caller-Network-Addr: 192.168.100.3 > Caller-Destination-Number: 444 > Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f > Caller-Source: mod_sofia > Caller-Context: mytest > Caller-Channel-Name: sofia/internal/699%40192.168.100.132 > Caller-Profile-Index: 1 > Caller-Profile-Created-Time: 1261278729764077 > Caller-Channel-Created-Time: 1261278729764077 > Caller-Channel-Answered-Time: 0 > Caller-Channel-Progress-Time: 0 > Caller-Channel-Progress-Media-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > > > sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f > call-command: execute > execute-app-name: answer > execute-app-arg: > > > Content-Type: command/reply > Reply-Text: +OK > > > sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f > call-command: execute > execute-app-name: playback > execute-app-arg: /tmp/ann.wav > > > Content-Type: command/reply > Reply-Text: +OK > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > > Sent: Saturday, December 19, 2009 5:30 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Difference between ESL execute() > > andexecuteAsync() > > > > I don't notice any different in behavior between execute() and > > executeAsync(). I was expecting that executeAsync() would return > > right-away, and that execute() would only return after the specified > > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > > > Running the sample app below, I see the "About to call execute(playback)" > > and "returned" displayed one right-after the other, even though the file > > being played takes about 4 minutes to play-out. > > > > Do I have this wrong, or is there something incorrect in my app? > > > > APP: > > #!/usr/bin/php > > > require_once "ESL.php"; > > > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > > $eventSocket->events('plain', 'CHANNEL_STATE'); > > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > > > // Wait for new call attempts > > while($eventSocket->connected()){ > > $event = $eventSocket->recvEvent(); > > $serializedBody = $event->serialize(); > > $listOfLines = toArrayOfLines($serializedBody); > > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > > > $uuid = $nameValuePairs['Caller-Unique-ID']; > > printf("New call from uuid: $uuid\n"); > > > > // answer the caller and play announcement > > $eventSocket->execute('answer', Null ,$uuid); > > > > printf("About to call execute(playback)\n"); > > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > > printf("returned\n"); > > } > > ?> > > > > > > DIALPLAN: > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > > This email was Anti Virus checked by Astaro Security Gateway. > > http://www.astaro.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/c369e10a/attachment-0002.html From andrew at hijacked.us Mon Dec 21 08:07:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 21 Dec 2009 11:07:08 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> Message-ID: <20091221160708.GC1956@hijacked.us> On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > Hi > > its good to hear > > any compare document between Vicidial and this project > No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew From jbr at consiglia.dk Mon Dec 21 08:13:21 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Mon, 21 Dec 2009 17:13:21 +0100 Subject: [Freeswitch-users] mod_xml_curl and gateways Message-ID: I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/ed0eb650/attachment-0002.html From mrene_lists at avgs.ca Mon Dec 21 08:21:04 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 21 Dec 2009 11:21:04 -0500 Subject: [Freeswitch-users] mod_xml_curl and gateways In-Reply-To: References: Message-ID: Hi, All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files. This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 21-Dec-09, at 11:13 AM, Jon Bruel wrote: > I wonder if it is possible to define common gateways (not user > specific gateways) by xml_curl, and if so, the bindings and syntax > to use? > > All the best /Jon > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/bec4ea70/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 21 08:22:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 10:22:54 -0600 Subject: [Freeswitch-users] How can I detect an execute failure using ESL? In-Reply-To: <0A22A70A58F642D0A3B778A11C17A67C@fromage> References: <0A22A70A58F642D0A3B778A11C17A67C@fromage> Message-ID: <191c3a030912210822x14699f7fid2e9e077c31102c8@mail.gmail.com> the latest version returns an event with that data in it similar to the api method. On Mon, Dec 21, 2009 at 2:48 AM, Ron McLeod wrote: > When I try and perform an operation on a channel which has gone, an error > is returned. How can I detect this using the ESL? execute() and sendRecv() > always return 0 (zero) regardless of whether the command returns *+OK* or > *?ERR*. > > > > sendmsg 5d09753c-ede7-11de-85c6-27ab474dd533 > > call-command: execute > > execute-app-name: hangup > > execute-app-arg: UNALLOCATED_NUMBER > > > > Content-Type: command/reply > > *Reply-Text: -ERR invalid session id > [5d09753c-ede7-11de-85c6-27ab474dd533]* > > > > Thanks, > > Ron > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/7b48361d/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 21 08:26:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Dec 2009 10:26:12 -0600 Subject: [Freeswitch-users] mod_xml_curl and gateways In-Reply-To: References: Message-ID: <191c3a030912210826k7d2ba4e5n400883b366d1fc4d@mail.gmail.com> same exact syntax only put the in the sofia profile On Mon, Dec 21, 2009 at 10:13 AM, Jon Bruel wrote: > I wonder if it is possible to define common gateways (not user specific > gateways) by xml_curl, and if so, the bindings and syntax to use? > > > > All the best /Jon > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/13312702/attachment-0002.html From freeswitch at aastral.net Mon Dec 21 08:47:17 2009 From: freeswitch at aastral.net (Bill W) Date: Mon, 21 Dec 2009 11:47:17 -0500 Subject: [Freeswitch-users] ACLs through proxy In-Reply-To: <4B2E9E1C.8090909@metik.com> References: <4B286906.7040502@aastral.net> <5ECB5D5F-B167-42C8-B928-FBEA28BFD3F0@freeswitch.org> <4B2A1902.2050008@aastral.net> <5ACAA1AE-832D-47AC-B739-C1347285A994@freeswitch.org> <4B2AC65F.5090806@aastral.net> <4B2AEC50.3030305@metik.com> <4B2AFEE8.5020002@aastral.net> <4B2B0342.3000201@metik.com> <4B2B2188.2060803@aastral.net> <4B2BBFA8.9050900@metik.com> <4B2C6694.3060400@aastral.net> <4B2D0222.7060609@metik.com> <4B2D5ECC.4060209@aastral.net> <4B2E9E1C.8090909@metik.com> Message-ID: <4B2FA695.3040803@aastral.net> Hey Metik, Thank you so much for your assistance on this issue. I really appreciate it. Yes I agree with you on the mod_xml_curl solution. However, as I was starting to pursue that, I ran into another issue. It appears as though I don't have access to any variables in the xml_curl POST that contain the IP of the UA. The only two variables with IPs (other than the switch IP) are: sip_contact_host=192.168.0.100 and ip=64.135.119.105 where the .105 is my proxy. :( Do you know of any way to get additional variables into the xml_curl POST? As far as my current use case, yes, you understand my needs correctly, with one slight modification, I want to use the IP acl+Auth with both REGISTERs and INVITEs. And yes, I agree with you that it is better to mitigate at the border, but I don't have that kind of infrastructure available yet. So do you have any other suggestions on a workaround with the xml_curl issue? Or should I include that with my bounty? Thanks, Bill Metik wrote: > Then it would appear that my original suggestion to use mod_xml_curl > would be best for now and you may need to offer a bounty for this > feature as others have suggested. Based on the sofia related snippets > presented--I would assume it would be trivial to implement since most of > the functionality is already there it just needs to be enhanced for your > purpose. It would also be extremely easy to do this in OpenSIPS as well > (using blacklists or avpops). > > Just so that I understand your dilemna, you want to reject an incoming > REGISTER associated with a specific user unless it comes from a fixed > location and if it does, you want to simply challenge it as usual to > prevent toll fraud? > > I have found that its best to mitigate an attack at ingress before it > even makes it to critical infrastructure (media gateways, > application/media servers, etc.). > > -metik > > Bill W. wrote: >> Hey Metik, >> >> Yes. Well, actually, I can have the cidr in two places in the directory. >> >> >> >> >> >> >From what I understand the cidr= parmeter is used in conjunction with >> the apply-inbound-acl parameter in the sofia profile to just allow >> someone to make calls from a certain IP without authenticating. >> >> And from what I understand the auth-acl= parameter is used to restrict a >> user to a particular cidr, but the user has to authenticate as well. >> >> *The second feature is the one I want to use.* I want to force users to >> authenticate, but only allow that authentication from a particular cidr >> as an added measure against toll fraud. >> >> And this appears to be causing the issue. Because once I specify the >> auth-acl parameter in the directory, sofia-reg enforces that acl. And >> unfortunately it's using the IP of the proxy, not of the user-agent. >> >> I looked in sofia.c and found this comment: >> /* >> * if network_ip is a proxy allowed to send calls, check for auth >> * ip header and see if it matches against the inbound acl >> */ >> >> And this coincides with my testing. >> I have in my >> profile. I have my proxy sending the X-AUTH-IP header (verified with >> tcpdump). And yet the REGISTER is still being denied. >> >> So it appears that the apply-proxy-acl is set up to work with the >> apply-inbound-acl ( to allow users from an IP without authenticating) >> >> But that hasn't been carried over to sofia_reg.c, which appears to >> simply check the IP of who FreeSWITCH is talking to against the auth-acl >> cidr specified in the directory. (Line 1926) >> >> So I guess the question is, is my analysis correct? >> >> Thoughts anyone? >> >> Thanks, >> Bill >> >> >> >> >> >> >> Metik wrote: >> >>> Bill, >>> >>> I think you would add this to the user profile in the directory. The >>> "brian.xml" example (located in ${confdir}/directory/) provided with the >>> default/sample configuration files demonstrates how to to do this by >>> introducing a "cidr" attribute to the the "user" element. >>> >>> Example: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> "http://wiki.freeswitch.org/wiki/Acl" contains some great info >>> (including a relevant example). >>> >>> -metik >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Mon Dec 21 09:02:33 2009 From: freeswitch at aastral.net (Bill W) Date: Mon, 21 Dec 2009 12:02:33 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> Message-ID: <4B2FAA29.4010405@aastral.net> I recently added an overview to this wiki page to help make things more clear as to which ACL you need for different purposes. http://wiki.freeswitch.org/wiki/ACL#Overview Thanks, Bill W. Mathieu Rene wrote: > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > It'll automatically add users with a cidr= attribute to the ACL list. > This way you can set channel variables in the users and use them through > your dialplan, all authenticated by ip address. > > Cheers, > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca From msc at freeswitch.org Mon Dec 21 09:48:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Dec 2009 09:48:44 -0800 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <20091221160708.GC1956@hijacked.us> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> <20091221160708.GC1956@hijacked.us> Message-ID: <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson wrote: > On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > > Hi > > > > its good to hear > > > > any compare document between Vicidial and this project > > > > No document, but briefly: > > * More focused on inbound than on outbound (at least for the moment) > vicidial is more geared for outbound. > * Handles email in queue (and soon chat), vicidial is only voice. > * wrapup time is per-call not static per-'campaign' > * license is a little more liberal > * can operate as a distributed system > * doesn't need asterisk ;) > Now *that* is a feature worth paying for! ;) Also, I thought you had a community edition vs. a professional edition? If so could you explain the difference? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/6afc69f1/attachment-0002.html From qinglan_zeng at hotmail.com Mon Dec 21 10:16:14 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Mon, 21 Dec 2009 18:16:14 +0000 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: Hello, I noticed some guys had develop the Skype module while there is a policy from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means)" , which means once you use Skype unlimited calls plan into PBX, Skype will frozen your account without any money return. That's a big risk for anybody to use Skype unlimited call plan. My question is how do we avoid such kind of risk? Thanks Daniel Zeng From: freeswitch-users-request at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 42, Issue 193 To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 08:21:08 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --??????-- From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 09:42:30 -0600 Subject: Re: [Freeswitch-users] Difference between ESL execute() andexecuteAsync() if you run the socket in async mode, every call to execute is async if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod wrote: Here's the ES network trace: Content-Length: 1502 Content-Type: text/event-plain Event-Name: CHANNEL_STATE Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-19%2019%3A12%3A09 Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT Event-Date-Timestamp: 1261278729767397 Event-Calling-File: switch_channel.c Event-Calling-Function: switch_channel_perform_set_running_state Event-Calling-Line-Number: 1024 Channel-State: CS_ROUTING Channel-State-Number: 2 Channel-Name: sofia/internal/699%40192.168.100.132 Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: ringing Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 699 Caller-Dialplan: XML Caller-Caller-ID-Name: Ron%20Soft%20Phone Caller-Caller-ID-Number: 699 Caller-Network-Addr: 192.168.100.3 Caller-Destination-Number: 444 Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f Caller-Source: mod_sofia Caller-Context: mytest Caller-Channel-Name: sofia/internal/699%40192.168.100.132 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1261278729764077 Caller-Channel-Created-Time: 1261278729764077 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: answer execute-app-arg: Content-Type: command/reply Reply-Text: +OK sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f call-command: execute execute-app-name: playback execute-app-arg: /tmp/ann.wav Content-Type: command/reply Reply-Text: +OK > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Saturday, December 19, 2009 5:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Difference between ESL execute() > andexecuteAsync() > > I don't notice any different in behavior between execute() and > executeAsync(). I was expecting that executeAsync() would return > right-away, and that execute() would only return after the specified > application runs to completion (CHANNEL_EXECUTE_COMPLETE event). > > Running the sample app below, I see the "About to call execute(playback)" > and "returned" displayed one right-after the other, even though the file > being played takes about 4 minutes to play-out. > > Do I have this wrong, or is there something incorrect in my app? > > APP: > #!/usr/bin/php > require_once "ESL.php"; > > $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon'); > $eventSocket->events('plain', 'CHANNEL_STATE'); > $eventSocket->filter('channel-state', 'CS_ROUTING'); > > // Wait for new call attempts > while($eventSocket->connected()){ > $event = $eventSocket->recvEvent(); > $serializedBody = $event->serialize(); > $listOfLines = toArrayOfLines($serializedBody); > $nameValuePairs = toArrayOfNameValuePairs($listOfLines); > > $uuid = $nameValuePairs['Caller-Unique-ID']; > printf("New call from uuid: $uuid\n"); > > // answer the caller and play announcement > $eventSocket->execute('answer', Null ,$uuid); > > printf("About to call execute(playback)\n"); > $eventSocket->execute('playback', '/tmp/ann.wav', $uuid); > printf("returned\n"); > } > ?> > > > DIALPLAN: > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 --??????-- From: andrew at hijacked.us To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 11:07:08 -0500 Subject: Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > Hi > > its good to hear > > any compare document between Vicidial and this project > No document, but briefly: * More focused on inbound than on outbound (at least for the moment) vicidial is more geared for outbound. * Handles email in queue (and soon chat), vicidial is only voice. * wrapup time is per-call not static per-'campaign' * license is a little more liberal * can operate as a distributed system * doesn't need asterisk ;) Andrew --??????-- From: jbr at consiglia.dk To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 17:13:21 +0100 Subject: [Freeswitch-users] mod_xml_curl and gateways I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon --??????-- From: mrene_lists at avgs.ca To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 11:21:04 -0500 Subject: Re: [Freeswitch-users] mod_xml_curl and gateways Hi, All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files. This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 21-Dec-09, at 11:13 AM, Jon Bruel wrote: I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use? All the best /Jon _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ ?Windows Live ???????Messenger2009???? http://www.windowslive.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/10f83f1c/attachment-0002.html From itamar at ispbrasil.com.br Mon Dec 21 10:37:38 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Mon, 21 Dec 2009 16:37:38 -0200 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: 2009/12/21 ??? : > Hello, > > I noticed some guys had develop the Skype module while there is a policy > from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used > by one person only and is not to be shared with any other user (whether via > a PBX, call centre, computer or any other means)" , which means once you use > Skype unlimited calls plan into PBX, Skype will frozen your account without > any money return. That's a big risk for anybody to use Skype unlimited call > plan. > > My question is how do we avoid such kind of risk? > > Thanks > Daniel Zeng the best answer is don't use skype. ------------ Itamar Reis Peixoto e-mail/msn/google talk/sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From brian at freeswitch.org Mon Dec 21 10:43:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 12:43:25 -0600 Subject: [Freeswitch-users] Skypiax: Skype account frozen In-Reply-To: References: Message-ID: So says the man with his Skype username in his sig! :P /b On Dec 21, 2009, at 12:37 PM, Itamar Reis Peixoto wrote: > the best answer is don't use skype. > > > > ------------ > > Itamar Reis Peixoto > > e-mail/msn/google talk/sip: itamar at ispbrasil.com.br > skype: itamarjp > icq: 81053601 > +55 11 4063 5033 > +55 34 3221 8599 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/041fac2e/attachment-0002.html From JCasale at activenetwerx.com Mon Dec 21 11:56:59 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 19:56:59 +0000 Subject: [Freeswitch-users] sound rpms Message-ID: So the spec from trunk says "Soundfiles are moving into a separate spec" but I can't find this spec anywhere in svn? Anyone know where it is? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/bfe3a72a/attachment-0002.html From mike at jerris.com Mon Dec 21 12:25:59 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Dec 2009 15:25:59 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: Message-ID: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Working on it, moving the repos around to do this right... http://jira.freeswitch.org/browse/FSBUILD-218 Mike On Dec 21, 2009, at 2:56 PM, Joseph L. Casale wrote: > So the spec from trunk says ?Soundfiles are moving into a separate spec? > but I can?t find this spec anywhere in svn? > > Anyone know where it is? > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/0a2de892/attachment-0002.html From JCasale at activenetwerx.com Mon Dec 21 12:49:28 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 20:49:28 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >Working on it, moving the repos around to do this right... > >http://jira.freeswitch.org/browse/FSBUILD-218 > >Mike Thanks, Is this known to not work with non root builds? It errored out after creating some messy hierarchies with the actual variable calls, instead of their values? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/91a52343/attachment-0002.html From mike at jerris.com Mon Dec 21 13:00:35 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 21 Dec 2009 16:00:35 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: This is a total work in progress that has not even merged into tree. So it is not "known" to work or not work anywhere. Patches to correct issues are welcome. Mike On Dec 21, 2009, at 3:49 PM, Joseph L. Casale wrote: > >Working on it, moving the repos around to do this right... > > > >http://jira.freeswitch.org/browse/FSBUILD-218 > > > >Mike > > Thanks, Is this known to not work with non root builds? It errored out after creating some > messy hierarchies with the actual variable calls, instead of their values? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/ba0d897f/attachment-0002.html From JCasale at activenetwerx.com Mon Dec 21 13:04:51 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 21 Dec 2009 21:04:51 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >Thanks, Is this known to not work with non root builds? It errored out after creating some >messy hierarchies with the actual variable calls, instead of their values? Actually, I tried on a lab vm as root in the typical dirs. and got the same result: ... `./us/callie/time/48000/hours.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/hours.wav' `./us/callie/time/48000/oclock.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/oclock.wav' `./us/callie/time/48000/mon-6.wav' -> `%{buildroot}/opt/freeswitch/sounds/en/us/callie/time/48000/mon-6.wav' error: Bad exit status from /var/tmp/rpm-tmp.19453 (%install) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.19453 (%install) From stevendt at primrosebank.net Mon Dec 21 13:25:36 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Mon, 21 Dec 2009 21:25:36 -0000 Subject: [Freeswitch-users] Building on Windows References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com><20091217171527.GA16380@hijacked.us> Message-ID: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Hi Mike, OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! Most of the "warnings" in the build have been cleared, I notice that the "error" in the build of mod_opal has gone with mod_opal not being in the preconfigured build list now and also that the number of warnings on VS2008 has reduced since I've been playing with this over the past few days and SVN versions. There does not seem to be anything major wrong, but as you requested, I have raised a Jira (FSBUILD-221) that you might care to take a look at please ? I attached a copy of the output from the build run and highlighted in bold in the RTF file the warnings that are generated. There are a host of warnings due to the use of /analyze which is appears not to be supported by the Express compiler. Most significant (though still trivial) are the warnings of some type conversion problems and some "indirection" errors. As I said, these don't seem to be too much of a problem, but you may care to take a look when you have time. One more thing, when I do subsequent builds, are there any pre-build steps that I need to take ? When I tried rebuilding previously, there seemed to be some directories that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) that were flagged as problems - do I just ignore these warnings or unprotect and/or delete the directories before rebuilding ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 17, 2009 6:14 PM Subject: Re: [Freeswitch-users] Building on Windows On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/fccae920/attachment-0002.html From andrew at hijacked.us Mon Dec 21 14:36:59 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 21 Dec 2009 17:36:59 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter) In-Reply-To: <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> References: <20091219014359.GA21798@hijacked.us> <20091219031649.GA1956@hijacked.us> <20091221160708.GC1956@hijacked.us> <87f2f3b90912210948h8362258ib2eb981ca38e43f3@mail.gmail.com> Message-ID: <20091221223659.GE1956@hijacked.us> On Mon, Dec 21, 2009 at 09:48:44AM -0800, Michael Collins wrote: > On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson wrote: > > > On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote: > > > Hi > > > > > > its good to hear > > > > > > any compare document between Vicidial and this project > > > > > > > No document, but briefly: > > > > * More focused on inbound than on outbound (at least for the moment) > > vicidial is more geared for outbound. > > * Handles email in queue (and soon chat), vicidial is only voice. > > * wrapup time is per-call not static per-'campaign' > > * license is a little more liberal > > * can operate as a distributed system > > * doesn't need asterisk ;) > > > Now *that* is a feature worth paying for! ;) > > Also, I thought you had a community edition vs. a professional edition? If > so could you explain the difference? I've managed to avoid that thus far, I suspect that something like an outbound campaign manager (which could be implemented as just another media type) might be something to fall under that sort of split, but right now the release includes everything we've got (including an integration module that's probably of limited use to anyone else - but its a good example of how to build your own). Andrew From larclap at yahoo.com Mon Dec 21 14:50:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Mon, 21 Dec 2009 14:50:11 -0800 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <4B2FAA29.4010405@aastral.net> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> Message-ID: <007701ca828f$f414f9b0$dc3eed10$@com> Bill, Thanks for your ACL Overview. Perhaps you can help me understand more clearly. If you include the "local-network-acl" and "apply-inbound-acl" params in the sip_profiles and setup the list for "localnet.auto" in acl.conf.xml, does this mean you do not have to include the cidr attribute for individual extensions in the directory/default folder? Is "apply-inbound-acl" supposed to exist in both internal and external profiles while "apply-inbound-acl" is only in the internal? Thanks, Lars > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users- > bounces at lists.freeswitch.org] On Behalf Of Bill W > Sent: Monday, December 21, 2009 9:03 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Authenticating end points by IP > > I recently added an overview to this wiki page to help make things more > clear as to which ACL you need for different purposes. > > http://wiki.freeswitch.org/wiki/ACL#Overview > > Thanks, > Bill W. > > > Mathieu Rene wrote: > > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > > > It'll automatically add users with a cidr= attribute to the ACL list. > > This way you can set channel variables in the users and use them through > > your dialplan, all authenticated by ip address. > > > > Cheers, > > > > Mathieu Rene > > Avant-Garde Solutions Inc > > Office: + 1 (514) 664-1044 x100 > > Cell: +1 (514) 664-1044 x200 > > mrene at avgs.ca > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Mon Dec 21 15:53:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Dec 2009 15:53:49 -0800 Subject: [Freeswitch-users] Setting Restrictions on Default Dialplan In-Reply-To: <26870380.post@talk.nabble.com> References: <26868725.post@talk.nabble.com> <26870380.post@talk.nabble.com> Message-ID: <87f2f3b90912211553r75204938y2ea845020db1ac01@mail.gmail.com> On Sun, Dec 20, 2009 at 10:31 PM, Edmar Cruz wrote: > > Where should I write this line > > > > data="misc/you-are-not-authorized.wav"/> > > > > > > > on the default.xml or in the default category? > You can put it in default.xml or in an xml file in conf/dialplan/default/ -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/08d5c351/attachment-0002.html From JCasale at activenetwerx.com Mon Dec 21 16:03:48 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Tue, 22 Dec 2009 00:03:48 +0000 Subject: [Freeswitch-users] Variables for install directories Message-ID: Searching through the wiki for any indication as to what if any variables exist for the install location in that I can leverage in a script. Can anyone point me along, I can't seem to find anything. I want to place a shell script in /opt/freeswitch/scripts that needs a reference to a conf file that a binary it runs is calling. So now I have in two places hardcoded paths that I was hoping to avoid, in the dialplan and in the shell script. When either of these is run, does there exist something like and the same for use inside the shell script? Thanks! jlc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/e5c757ef/attachment-0002.html From jerry.richards at teotech.com Mon Dec 21 16:24:06 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 21 Dec 2009 16:24:06 -0800 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> Message-ID: <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the WARNING in the Freeswitch log. However, it still behaves the same way. That is, the internal callee rings for about 12 seconds, then stops ringing, and the PSTN caller just hears ringback for about 60 seconds and is not given the opportunity to leave voice mail. In contrast, an internal-to-internal call will go to voice mail after 30 seconds. I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe driver (or Freeswitch) setting that would correct this? Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, November 03, 2009 2:23 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] WARNING On Inbound Call Question can you try the same thing with the latest trunk or pre-release tarball. On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards wrote: I have my Freeswitch server with an installed Sangoma A101D card. Most everything works okay, however, when I get an inbound call from the PSTN, I see the following warning show up in the log. Additionally, the caller (on the PSTN) does not hear ringback, and if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. Here are the two warnings: [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA Here is the log of the warning upon an inbound call: freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> 2009-11-02 09:06:01.664835 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0 Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176] 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on 1:1 from DOWN to RING 2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig [START] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound channel OpenZAP/1:1/5384 2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132] 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_INIT 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/5384) State INIT going to sleep 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING 2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384 CHANNEL ROUTING 2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78 OpenZAP/1:1/5384 Standard ROUTING 2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing 4253813176->5384 in context default Dialplan: OpenZAP/1:1/5384 parsing [default->unloop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->tod_example] continue=true Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example] Dialplan: OpenZAP/1:1/5384 Action set(open=true) Dialplan: OpenZAP/1:1/5384 parsing [default->SangomaPRI] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI] destination_number(5384) =~ /^9(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept] destination_number(5384) =~ /^(5380)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->group-intercept] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept] destination_number(5384) =~ /^\*8$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->intercept-ext] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext] destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->redial] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~ /^870$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->global] continue=true Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: OpenZAP/1:1/5384 Absolute Condition [global] Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-2] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-2] destination_number(5384) =~ /^9001$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->snom-demo-1] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [snom-demo-1] destination_number(5384) =~ /^9000$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->eavesdrop] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [eavesdrop] destination_number(5384) =~ /^779$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call_return] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call_return] destination_number(5384) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->del-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [del-group] destination_number(5384) =~ /^80(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->add-group] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [add-group] destination_number(5384) =~ /^81(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-simo] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-simo] destination_number(5384) =~ /^82(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->call-group-order] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [call-group-order] destination_number(5384) =~ /^83(\d{2})$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->extension-intercom] continue=false Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [extension-intercom] destination_number(5384) =~ /^8(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 parsing [default->Local_Extension] continue=false Dialplan: OpenZAP/1:1/5384 Regex (PASS) [Local_Extension] destination_number(5384) =~ /^(5[34][8901][0-9])$/ break=on-false Dialplan: OpenZAP/1:1/5384 Action set(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action export(dialed_extension=5384) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: OpenZAP/1:1/5384 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: OpenZAP/1:1/5384 Action set(ringback=${us-ring}) Dialplan: OpenZAP/1:1/5384 Action set(transfer_ringback=local_stream://moh) Dialplan: OpenZAP/1:1/5384 Action set(call_timeout=30) Dialplan: OpenZAP/1:1/5384 Action set(hangup_after_bridge=true) Dialplan: OpenZAP/1:1/5384 Action set(continue_on_fail=true) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: OpenZAP/1:1/5384 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: OpenZAP/1:1/5384 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: OpenZAP/1:1/5384 Action answer() Dialplan: OpenZAP/1:1/5384 Action sleep(1000) Dialplan: OpenZAP/1:1/5384 Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:114 (OpenZAP/1:1/5384) State Change CS_ROUTING -> CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/5384) State ROUTING going to sleep 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/5384) Running State Change CS_EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:491 (OpenZAP/1:1/5384) State EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] mod_openzap.c:408 OpenZAP/1:1/5384 CHANNEL EXECUTE 2009-11-02 09:06:01.666685 [DEBUG] switch_core_state_machine.c:151 OpenZAP/1:1/5384 Standard EXECUTE EXECUTE OpenZAP/1:1/5384 set(open=true) 2009-11-02 09:06:01.666685 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [open]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-spymap/4253813176/b678f311-ab74-4cc1-afac-b83d89a 53132) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/4253813176/5384) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/global/b678f311-ab74-4cc1-afac-b83d89a5 3132) EXECUTE OpenZAP/1:1/5384 set(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 export(dialed_extension=5384) 2009-11-02 09:06:01.667682 [DEBUG] mod_dptools.c:886 EXPORT [dialed_extension]=[5384] EXECUTE OpenZAP/1:1/5384 bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-02 09:06:01.667682 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE OpenZAP/1:1/5384 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/4253813176.2009-11-02-09-06 -01.wav EXECUTE OpenZAP/1:1/5384 bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-02 09:06:01.668708 [INFO] switch_ivr_async.c:1795 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE OpenZAP/1:1/5384 set(ringback=%(2000,4000,440.0,480.0)) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE OpenZAP/1:1/5384 set(transfer_ringback=local_stream://moh) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [transfer_ringback]=[local_stream://moh] EXECUTE OpenZAP/1:1/5384 set(call_timeout=30) 2009-11-02 09:06:01.668708 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [call_timeout]=[30] EXECUTE OpenZAP/1:1/5384 set(hangup_after_bridge=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [hangup_after_bridge]=[true] EXECUTE OpenZAP/1:1/5384 set(continue_on_fail=true) 2009-11-02 09:06:01.669681 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [continue_on_fail]=[true] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-call_return/5384/4253813176) EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial_ext/5384/b678f311-ab74-4cc1-afac-b83d89 a53132) EXECUTE OpenZAP/1:1/5384 set(called_party_callgroup=techsupport) 2009-11-02 09:06:01.670679 [DEBUG] mod_dptools.c:748 OpenZAP/1:1/5384 SET [called_party_callgroup]=[techsupport] EXECUTE OpenZAP/1:1/5384 hash(insert/192.168.72.141-last_dial/techsupport/b678f311-ab74-4cc1-afac-b83 d89a53132) EXECUTE OpenZAP/1:1/5384 bridge(user/5384 at 192.168.72.141) 2009-11-02 09:06:01.671683 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=5384 at 192.168.72.141] 2009-11-02 09:06:01.671683 [NOTICE] switch_channel.c:602 New Channel sofia/internal/sip:5384 at 192.168.72.163:5060 [9e7b8fae-6194-430c-951b-948ebd2c2a3b] 2009-11-02 09:06:01.671683 [DEBUG] mod_sofia.c:2811 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_NEW -> CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_INIT 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:83 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA INIT 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_INIT -> CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/sip:5384 at 192.168.72.163:5060) State INIT going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING 2009-11-02 09:06:01.672688 [DEBUG] mod_sofia.c:130 sofia/internal/sip:5384 at 192.168.72.163:5060 SOFIA ROUTING 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/sip:5384 at 192.168.72.163:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:5384 at 192.168.72.163:5060 [BREAK] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/sip:5384 at 192.168.72.163:5060) State ROUTING going to sleep 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/sip:5384 at 192.168.72.163:5060) Running State Change CS_CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [calling][0] 2009-11-02 09:06:01.672688 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/sip:5384 at 192.168.72.163:5060) State CONSUME_MEDIA 2009-11-02 09:06:01.672688 [DEBUG] switch_ivr_originate.c:1701 OpenZAP/1:1/5384 receive message [PROGRESS] 2009-11-02 09:06:01.673742 [DEBUG] mod_openzap.c:759 Changing state on 1:1 from RING to PROGRESS 2009-11-02 09:06:01.674787 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [PROGRESS] 2009-11-02 09:06:01.675844 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=0 Seq=11 2009-11-02 09:06:01.684776 [WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to PROGRESS_MEDIA 2009-11-02 09:06:01.684776 [DEBUG] switch_core_session.c:630 Send signal OpenZAP/1:1/5384 [BREAK] 2009-11-02 09:06:01.684776 [NOTICE] switch_ivr_originate.c:1701 Pre-Answer OpenZAP/1:1/5384! 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-11-02 09:06:01.684776 [DEBUG] switch_ivr_originate.c:1777 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-11-02 09:06:01.693835 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:5384 at 192.168.72.163:5060 entering state [proceeding][180] 2009-11-02 09:06:01.693835 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:5384 at 192.168.72.163:5060! 2009-11-02 09:06:01.705777 [DEBUG] switch_core_io.c:649 OpenZAP/1:1/5384 receive message [TRANSCODING_NECESSARY] freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> freeswitch at TeoProxy.greyhawk.tonecommander.com> Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/4052d79c/attachment-0002.html From jerry.richards at teotech.com Mon Dec 21 16:44:49 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 21 Dec 2009 16:44:49 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time Message-ID: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry From brian at freeswitch.org Mon Dec 21 16:52:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Dec 2009 18:52:20 -0600 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> Message-ID: <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> You know that warning is meaningless. Search the archives we have talked about this to no end it seems. And I'm sure Moy fixed this. /b On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: > Okay, I upgraded to 1.0.5pre9 and tried this test again and I do not see the WARNING in the Freeswitch log. However, it still behaves the same way. That is, the internal callee rings for about 12 seconds, then stops ringing, and the PSTN caller just hears ringback for about 60 seconds and is not given the opportunity to leave voice mail. In contrast, an internal-to-internal call will go to voice mail after 30 seconds. > > I put a new 11595 log into the pastebin. Is there some Sangoma Wanpipe driver (or Freeswitch) setting that would correct this? > > Best Regards, > Jerry > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/05e3af79/attachment-0002.html From jeff at jefflenk.com Mon Dec 21 20:18:40 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 22:18:40 -0600 Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com><20091217171527.GA16380@hijacked.us>, , <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff From: stevendt at primrosebank.net To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Dec 2009 21:25:36 +0000 Subject: Re: [Freeswitch-users] Building on Windows Hi Mike, OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! Most of the "warnings" in the build have been cleared, I notice that the "error" in the build of mod_opal has gone with mod_opal not being in the preconfigured build list now and also that the number of warnings on VS2008 has reduced since I've been playing with this over the past few days and SVN versions. There does not seem to be anything major wrong, but as you requested, I have raised a Jira (FSBUILD-221) that you might care to take a look at please ? I attached a copy of the output from the build run and highlighted in bold in the RTF file the warnings that are generated. There are a host of warnings due to the use of /analyze which is appears not to be supported by the Express compiler. Most significant (though still trivial) are the warnings of some type conversion problems and some "indirection" errors. As I said, these don't seem to be too much of a problem, but you may care to take a look when you have time. One more thing, when I do subsequent builds, are there any pre-build steps that I need to take ? When I tried rebuilding previously, there seemed to be some directories that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) that were flagged as problems - do I just ignore these warnings or unprotect and/or delete the directories before rebuilding ? regards Dave ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 17, 2009 6:14 PM Subject: Re: [Freeswitch-users] Building on Windows On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: Hi, I'm probably going to regret this - I'm not sure that I'll be able to do this without a lot of pain (nothing to do with FS - more my lack of ability with Visual Studio), but......, I want to try building FreeSwitch from source rather than using the pre-built binaries. I have a couple of initial questions that, hopefully, someone can answer please ? 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 on the horizon for me. Having downloaded the SVN, I see there is a VS 2005 Solution, but it is marked as "Unsupported", although the Wiki says that you only need VC++2005. What does "unsupported" mean in this context ? I guess that support for VS2005 is being dropped, but is the VS2005 Solution still being maintained, and if so, for how long? I'd hate to get into the build thing and then find that I was stalled when VS2005 support was dropped altogether ? Install VS 2008 if at all possible (express edition is free). 2005 support isn't maintained much if at all, so a lot of newer modules stand a good chance of not having support. We maintain it as far as things that work now shouldn't break, but we rarely test it and only fix things when people supply patches or let me know there is a problem so I can address it. 2. The whole SVN thing is new to me but I've worked out that I need an SVN Client on Windows to work with the source. Can anyone recommend the best (free) SVN Client for Windows to use with FreeSwitch. I have installed TortoiseSVN - a Windows Explorer Shell that looks pretty and seemed to work on my first build but it's not command line based so some of the tips given in the Wiki like "make current" and "make sounds" may be more awkward to achieve. Is anyone else using Tortoise and/or can give some tips on which SVN client to use ? Tortoise SVN is fine and is probably the de-facto client for windows. make current and such are all for the unix build only, on the msvc (at least 2008) build they are all built right into the solution ] 3. I built 15979 last night (with VS2005) and got some warnings, with data type conversion - is this a known issue under Windows ? 2005 has slightly different warning settings than are even available in 2008 so I get these from time to time. If you open up a bug on jira.freeswitch.org for me with details I can try to get them corrected. 4. There was one fatal error in the build of mod_opal (missing file) (Some examples of the warnings and the error are shown below :-) Try with VS 2008 and see if they go away. I think this is due to missing dependencies. I don't think I had automation to download the right svn versions of opal. 5. How do I specify which options (e.g., mod_flite, to be included iin the build. You can enable the different sub projects somehow in the UI, I always forget exactly how but just click around in VS and you'll find it. You can adjust this in the configuration managaer 6. How do I build the sounds etc. ? The sounds are a subproject too IIRC. I think think might only be in the 2008 versions, I can't recall to be sure, but there are targets you can build that will install them. Mike _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Hotmail: Free, trusted and rich email service. http://clk.atdmt.com/GBL/go/171222984/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091221/54ad2aca/attachment-0002.html From jeff at jefflenk.com Mon Dec 21 20:22:09 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 20:22:09 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: <1261455729581-4201954.post@n2.nabble.com> Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff Dave Stevenson wrote: > > Hi Mike, > > OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! > > Most of the "warnings" in the build have been cleared, I notice that the > "error" in the build of mod_opal has gone with mod_opal not being in the > preconfigured build list now and also that the number of warnings on > VS2008 has reduced since I've been playing with this over the past few > days and SVN versions. > > There does not seem to be anything major wrong, but as you requested, I > have raised a Jira (FSBUILD-221) that you might care to take a look at > please ? > > I attached a copy of the output from the build run and highlighted in bold > in the RTF file the warnings that are generated. > There are a host of warnings due to the use of /analyze which is appears > not to be supported by the Express compiler. > > Most significant (though still trivial) are the warnings of some type > conversion problems and some "indirection" errors. > > As I said, these don't seem to be too much of a problem, but you may care > to take a look when you have time. > > One more thing, when I do subsequent builds, are there any pre-build steps > that I need to take ? > When I tried rebuilding previously, there seemed to be some directories > that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) > that were flagged as problems - do I just ignore these warnings or > unprotect and/or delete the directories before rebuilding ? > > regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, December 17, 2009 6:14 PM > Subject: Re: [Freeswitch-users] Building on Windows > > > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > > > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > > Hi, > > > > I'm probably going to regret this - I'm not sure that I'll be able > to do this without a lot of pain (nothing to do with FS - more my lack of > ability with Visual Studio), but......, I want to try building FreeSwitch > from source rather than using the pre-built binaries. I have a couple of > initial questions that, hopefully, someone can answer please ? > > > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 > on the horizon for me. > > Having downloaded the SVN, I see there is a VS 2005 Solution, but it > is marked as "Unsupported", although the Wiki says that you only need > VC++2005. > > What does "unsupported" mean in this context ? I guess that support > for VS2005 is being dropped, but is the VS2005 Solution still being > maintained, and if so, for how long? I'd hate to get into the build thing > and then find that I was stalled when VS2005 support was dropped > altogether ? > > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules > stand > a good chance of not having support. > > > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > > > > > 2. The whole SVN thing is new to me but I've worked out that I need > an SVN Client on Windows to work with the source. Can anyone recommend the > best (free) SVN Client for Windows to use with FreeSwitch. I have > installed TortoiseSVN - a Windows Explorer Shell that looks pretty and > seemed to work on my first build but it's not command line based so some > of the tips given in the Wiki like "make current" and "make sounds" may be > more awkward to achieve. Is anyone else using Tortoise and/or can give > some tips on which SVN client to use ? > > > > Tortoise SVN is fine and is probably the de-facto client for windows. > > > > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] > > 3. I built 15979 last night (with VS2005) and got some warnings, > with data type conversion - is this a known issue under Windows ? > > > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > > > > > 4. There was one fatal error in the build of mod_opal (missing file) > > (Some examples of the warnings and the error are shown below :-) > > > > Try with VS 2008 and see if they go away. > > > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > > > 5. How do I specify which options (e.g., mod_flite, to be included > iin the build. > > > > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. > > > > You can adjust this in the configuration managaer > > > 6. How do I build the sounds etc. ? > > > > > The sounds are a subproject too IIRC. > > > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > > > Mike > > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4201954.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Mon Dec 21 20:23:31 2009 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 21 Dec 2009 20:23:31 -0800 (PST) Subject: [Freeswitch-users] Building on Windows In-Reply-To: <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> References: <9CF78914970D4FC39B275EC0BABB56C9@bp1.ad.bp.com> <20091217171527.GA16380@hijacked.us> <9EE4B59BFBD8477B86C7A75D907FE7A1@bp1.ad.bp.com> Message-ID: <1261455811463-4201959.post@n2.nabble.com> Hi Dave, I have corrected several of the warnings. On subsequent builds the download errors can be ignored(files already present). Jeff Dave Stevenson wrote: > > Hi Mike, > > OK - have "bitten the bullet" and installed VS2008 Express over VS2005 ! > > Most of the "warnings" in the build have been cleared, I notice that the > "error" in the build of mod_opal has gone with mod_opal not being in the > preconfigured build list now and also that the number of warnings on > VS2008 has reduced since I've been playing with this over the past few > days and SVN versions. > > There does not seem to be anything major wrong, but as you requested, I > have raised a Jira (FSBUILD-221) that you might care to take a look at > please ? > > I attached a copy of the output from the build run and highlighted in bold > in the RTF file the warnings that are generated. > There are a host of warnings due to the use of /analyze which is appears > not to be supported by the Express compiler. > > Most significant (though still trivial) are the warnings of some type > conversion problems and some "indirection" errors. > > As I said, these don't seem to be too much of a problem, but you may care > to take a look when you have time. > > One more thing, when I do subsequent builds, are there any pre-build steps > that I need to take ? > When I tried rebuilding previously, there seemed to be some directories > that could not be overwritten (e.g., \libs\pthreads-w32--2-7-0-release) > that were flagged as problems - do I just ignore these warnings or > unprotect and/or delete the directories before rebuilding ? > > regards > Dave > > ----- Original Message ----- > From: Michael Jerris > To: freeswitch-users at lists.freeswitch.org > Sent: Thursday, December 17, 2009 6:14 PM > Subject: Re: [Freeswitch-users] Building on Windows > > > > > On Dec 17, 2009, at 12:15 PM, Andrew Thompson wrote: > > > On Thu, Dec 17, 2009 at 05:02:10PM -0000, Dave Stevenson wrote: > > Hi, > > > > I'm probably going to regret this - I'm not sure that I'll be able > to do this without a lot of pain (nothing to do with FS - more my lack of > ability with Visual Studio), but......, I want to try building FreeSwitch > from source rather than using the pre-built binaries. I have a couple of > initial questions that, hopefully, someone can answer please ? > > > > 1. I only have Visual Studio 2005 and don't see an upgrade to 2008 > on the horizon for me. > > Having downloaded the SVN, I see there is a VS 2005 Solution, but it > is marked as "Unsupported", although the Wiki says that you only need > VC++2005. > > What does "unsupported" mean in this context ? I guess that support > for VS2005 is being dropped, but is the VS2005 Solution still being > maintained, and if so, for how long? I'd hate to get into the build thing > and then find that I was stalled when VS2005 support was dropped > altogether ? > > > Install VS 2008 if at all possible (express edition is free). 2005 > support isn't maintained much if at all, so a lot of newer modules > stand > a good chance of not having support. > > > > We maintain it as far as things that work now shouldn't break, but we > rarely test it and only fix things when people supply patches or let me > know there is a problem so I can address it. > > > > > 2. The whole SVN thing is new to me but I've worked out that I need > an SVN Client on Windows to work with the source. Can anyone recommend the > best (free) SVN Client for Windows to use with FreeSwitch. I have > installed TortoiseSVN - a Windows Explorer Shell that looks pretty and > seemed to work on my first build but it's not command line based so some > of the tips given in the Wiki like "make current" and "make sounds" may be > more awkward to achieve. Is anyone else using Tortoise and/or can give > some tips on which SVN client to use ? > > > > Tortoise SVN is fine and is probably the de-facto client for windows. > > > > > make current and such are all for the unix build only, on the msvc (at > least 2008) build they are all built right into the solution > ] > > 3. I built 15979 last night (with VS2005) and got some warnings, > with data type conversion - is this a known issue under Windows ? > > > > 2005 has slightly different warning settings than are even available in > 2008 so I get these from time to time. If you open up a bug on > jira.freeswitch.org for me with details I can try to get them corrected. > > > > > 4. There was one fatal error in the build of mod_opal (missing file) > > (Some examples of the warnings and the error are shown below :-) > > > > Try with VS 2008 and see if they go away. > > > > I think this is due to missing dependencies. I don't think I had > automation to download the right svn versions of opal. > > > 5. How do I specify which options (e.g., mod_flite, to be included > iin the build. > > > > You can enable the different sub projects somehow in the UI, I always > forget exactly how but just click around in VS and you'll find it. > > > > You can adjust this in the configuration managaer > > > 6. How do I build the sounds etc. ? > > > > > The sounds are a subproject too IIRC. > > > > I think think might only be in the 2008 versions, I can't recall to be > sure, but there are targets you can build that will install them. > > > > > Mike > > > > > ------------------------------------------------------------------------------ > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Building-on-Windows-tp4182382p4201959.html Sent from the freeswitch-users mailing list archive at Nabble.com. From john at acsol.net Mon Dec 21 16:15:22 2009 From: john at acsol.net (john at acsol.net) Date: Mon, 21 Dec 2009 17:15:22 -0700 Subject: [Freeswitch-users] Multitenant dialplans Message-ID: <4b300f9a.313.2c10.1142196461@acsol.net> I have Freeswitch setup and working as a single tenant system mostly using the default configuration. Trying to convert to a multitenant environment, I have used both the Multi-tenant and Multiple Companies wiki's. I get the phone to register, can call out using the external profile to a ITSP, can call music on hold; however I can not call other users in the company. It appears that when logged in with single company and default context it sucessfully calls other internal phones with bridge to "sofia/internal/sip:extersion at public-IP:translated-port"; however when I log into "Company1" with the phones, it tries "sofia/internal/dialed-extension at Company1" ... I also get "User not Registered". The dialplans are the same either way. Any ideas? Thanks John From Prometheus001 at gmx.net Tue Dec 22 03:20:39 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Dec 2009 12:20:39 +0100 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time In-Reply-To: References: Message-ID: <4B30AB87.3060909@gmx.net> Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a 200 > OK with only the PCMU codec. This looks incorrect. The audio call > previously negotiated to the speex/16000 codec, and the re-INVITE from the > caller added the H263-1998 codec. If I re-attempt to start video at the > caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the complete > scenario: establishing audio call, first failed start video attempt, and > second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Tue Dec 22 03:40:11 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 22 Dec 2009 12:40:11 +0100 Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf Message-ID: <4B30B01B.30809@gmx.net> Hello, in a bigger installation with some thousand endpoints in the field we see, that the endpoint equipment is always using INFO messages (standard setting is auto, so the endpoint decides which method to use). I have 2 questions to that scenario: 1. Is there a way that Freeswitch forces/restricts the endpoint to use rfc2833 or not to send to allow INFO in the invite message? 2. Currently INFO messages do not get forwarded from the caller through freeswitch to called endpoint. How can we enable that FS is fowarding the INFO messages? Best regards Peter From scott.torr.fs at letterboxes.org Tue Dec 22 06:57:05 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Wed, 23 Dec 2009 01:57:05 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? Message-ID: <1261493825.21085.1351311647@webmail.messagingengine.com> ubuntu-8.04.3-server-amd64.iso (update/upgrade) FreeSWITCH Version 1.0.trunk (15787) skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb mod_skypiax (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) fs>console loglevel 7 If I dial 501 from from a sip phone using "inband" dtmf I can see the dtmf tones being detected and decoded by fs in the debug log. If however I use a pstn phone and dial my skypeIN telephone number the call comes into fs via skypiax but when I generate dtmf tones on the phone they are not detected or decoded by fs. If I take the record_session file and spectrum analyze the recorded tones appear to be within spec. Can anybody suggest why this is not working for me? Is the correct sample rate being used in libteletone_detect.c? Does the Goertzel algorithm work for other sample rates other than 8000hz? I'm not sure why I can not get this to work? regards, Scott Torr From anthony.minessale at gmail.com Tue Dec 22 07:12:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Dec 2009 09:12:30 -0600 Subject: [Freeswitch-users] Adding H263 Video to Existing Call Fails First Time In-Reply-To: References: Message-ID: <191c3a030912220712o5d028687hb3be922eb56a47f6@mail.gmail.com> Can you repeat that same trace with latest trunk? On Mon, Dec 21, 2009 at 6:44 PM, Jerry Richards wrote: > > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a 200 > OK with only the PCMU codec. This looks incorrect. The audio call > previously negotiated to the speex/16000 codec, and the re-INVITE from the > caller added the H263-1998 codec. If I re-attempt to start video at the > caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the complete > scenario: establishing audio call, first failed start video attempt, and > second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/99bcf8eb/attachment-0002.html From brian at freeswitch.org Tue Dec 22 07:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 09:13:54 -0600 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4b300f9a.313.2c10.1142196461@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> Message-ID: <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. /b On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: > I have Freeswitch setup and working as a single tenant > system mostly using the default configuration. Trying to > convert to a multitenant environment, I have used both the > Multi-tenant and Multiple Companies wiki's. I get the phone > to register, can call out using the external profile to a > ITSP, can call music on hold; however I can not call other > users in the company. > It appears that when logged in with single company and > default context it sucessfully calls other internal phones > with bridge to > "sofia/internal/sip:extersion at public-IP:translated-port"; > however when I log into "Company1" with the phones, it tries > "sofia/internal/dialed-extension at Company1" ... I also get > "User not Registered". The dialplans are the same either > way. > > Any ideas? > > Thanks > John > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Dec 22 07:21:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Dec 2009 09:21:16 -0600 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261493825.21085.1351311647@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> Message-ID: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> add "start_dtmf" app to your dialplan before bridge to start the inband dtmf detector. On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr wrote: > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > FreeSWITCH Version 1.0.trunk (15787) > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > mod_skypiax > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > fs>console loglevel 7 > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > dtmf tones being detected and decoded by fs in the debug log. > > > If however I use a pstn phone and dial my skypeIN telephone number the > call comes into fs via skypiax but when I generate dtmf tones on the > phone they are not detected or decoded by fs. > > If I take the record_session file and spectrum analyze the recorded > tones appear to be within spec. > > > Can anybody suggest why this is not working for me? > > > Is the correct sample rate being used in libteletone_detect.c? > Does the Goertzel algorithm work for other sample rates other than > 8000hz? > > > I'm not sure why I can not get this to work? > > > > regards, > Scott Torr > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/e5a9da29/attachment-0002.html From gmaruzz at celliax.org Tue Dec 22 07:25:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 22 Dec 2009 16:25:21 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261493825.21085.1351311647@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> Message-ID: <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> It is probably because mod_skypiax does not analize incoming audio looking for dtmf, because the "normal" call from a Skype client peer sends *both* inband and out of band (signaling) dtmf. So, I choose to only detect out of band (signaling) dtmfs, and ignore possible inband dtmfs (in the audio flow), so to have the most reliable source (signaling) and spare cpu (not analizing the incoming audio). Never tought you can receive calls from skypeIN with inband dtmfs... Open a Jira for this, I'll think about. Also, let me know your toughts... -giovanni On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr wrote: > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > FreeSWITCH Version 1.0.trunk (15787) > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > mod_skypiax > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > ? > ? ? > ? ? ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > ? ? > ? > > > > fs>console loglevel 7 > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > dtmf tones being detected and decoded by fs in the debug log. > > > If however I use a pstn phone and dial my skypeIN telephone number the > call comes into fs via skypiax but when I generate dtmf tones on the > phone they are not detected or decoded by fs. > > If I take the record_session file and spectrum analyze the recorded > tones appear to be within spec. > > > Can anybody suggest why this is not working for me? > > > Is the correct sample rate being used in libteletone_detect.c? > Does the Goertzel algorithm work for other sample rates other than > 8000hz? > > > I'm not sure why I can not get this to work? > > > > regards, > Scott Torr > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Tue Dec 22 07:26:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 22 Dec 2009 16:26:01 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> Message-ID: <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> do as anthm say :-) On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale wrote: > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf > detector. > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > wrote: >> >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> FreeSWITCH Version 1.0.trunk (15787) >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> mod_skypiax >> >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> >> >> ? >> ? ? >> ? ?> >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> ? ? >> ? >> >> >> >> fs>console loglevel 7 >> >> >> If I dial 501 from from a sip phone using "inband" dtmf I can see the >> dtmf tones being detected and decoded by fs in the debug log. >> >> >> If however I use a pstn phone and dial my skypeIN telephone number the >> call comes into fs via skypiax but when I generate dtmf tones on the >> phone they are not detected or decoded by fs. >> >> If I take the record_session file and spectrum analyze the recorded >> tones appear to be within spec. >> >> >> Can anybody suggest why this is not working for me? >> >> >> Is the correct sample rate being used in libteletone_detect.c? >> Does the Goertzel algorithm work for other sample rates other than >> 8000hz? >> >> >> I'm not sure why I can not get this to work? >> >> >> >> regards, >> Scott Torr >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From jerry.richards at teotech.com Tue Dec 22 08:02:21 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 22 Dec 2009 08:02:21 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail Message-ID: <19FBB8C038E64C1DB92B2842AF8BCF01@greyhawk.tonecommander.com> I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry From john at acsol.net Tue Dec 22 08:16:13 2009 From: john at acsol.net (John) Date: Tue, 22 Dec 2009 09:16:13 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> Message-ID: <4B30F0CD.8040703@acsol.net> Thanks Brian. I did have both force-register-domain and force-register-db-domain commented in both the internal.xml and internal-ipv6.xml. The phones appear to register to the company1 domain, as shown in sofia status profile company1; however I have noticed that when I try to make a call to another a phone in the same domain, the system is trying to call sofia/internal/1004 at company1 -- this is when we get the message, user not registered. If I can the phones to just register to the IP address of the machine, they call fine and is shows sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan problem? In both cases I am just using the sample dialplan. On 12/22/2009 8:13 AM, Brian West wrote: > The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. > > /b > > On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: > > >> I have Freeswitch setup and working as a single tenant >> system mostly using the default configuration. Trying to >> convert to a multitenant environment, I have used both the >> Multi-tenant and Multiple Companies wiki's. I get the phone >> to register, can call out using the external profile to a >> ITSP, can call music on hold; however I can not call other >> users in the company. >> It appears that when logged in with single company and >> default context it sucessfully calls other internal phones >> with bridge to >> "sofia/internal/sip:extersion at public-IP:translated-port"; >> however when I log into "Company1" with the phones, it tries >> "sofia/internal/dialed-extension at Company1" ... I also get >> "User not Registered". The dialplans are the same either >> way. >> >> Any ideas? >> >> Thanks >> John >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jerry.richards at teotech.com Tue Dec 22 08:33:00 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 22 Dec 2009 08:33:00 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time In-Reply-To: <4B30AB87.3060909@gmx.net> References: <4B30AB87.3060909@gmx.net> Message-ID: No. The following lines is commented out (internal.xml): Thanks, Jerry -----Original Message----- From: Peter P GMX [mailto:Prometheus001 at gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a > 200 OK with only the PCMU codec. This looks incorrect. The audio > call previously negotiated to the speex/16000 codec, and the re-INVITE > from the caller added the H263-1998 codec. If I re-attempt to start > video at the caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the > complete > scenario: establishing audio call, first failed start video attempt, > and second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From brian at freeswitch.org Tue Dec 22 10:04:30 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 12:04:30 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] "a1-has" param in gateway setting In-Reply-To: <179369.24879.qm@web110203.mail.gq1.yahoo.com> References: <179369.24879.qm@web110203.mail.gq1.yahoo.com> Message-ID: I'm not too sure you can put an a1-hash on outbound auth. /b On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote: > Hi, > > Does any body know or has test the "a1-hash" parameter with gateway > setting? I am not sure if it is even allowed. I have the following > gateway setting but when the freeswitch starts up it simply ignores this > provider without any error message or attempt to register in the log > file. Thank you for your help in advance. > > > > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/1f3a4443/attachment-0002.html From john at acsol.net Tue Dec 22 10:21:32 2009 From: john at acsol.net (John) Date: Tue, 22 Dec 2009 11:21:32 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B30F0CD.8040703@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> Message-ID: <4B310E2C.8020501@acsol.net> One point of clarification, currently all the phones are behind NAT, so it appears that when the phones are in a Non-multitenant scenario, they use SIP:dialed_number at IP-address-of-their-gateway. On 12/22/2009 9:16 AM, John wrote: > Thanks Brian. I did have both force-register-domain and > force-register-db-domain commented in both the internal.xml and > internal-ipv6.xml. The phones appear to register to the company1 domain, > as shown in sofia status profile company1; however I have noticed that > when I try to make a call to another a phone in the same domain, the > system is trying to call sofia/internal/1004 at company1 -- this is when we > get the message, user not registered. If I can the phones to just > register to the IP address of the machine, they call fine and is shows > sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan > problem? In both cases I am just using the sample dialplan. > > > > > On 12/22/2009 8:13 AM, Brian West wrote: > >> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. >> >> /b >> >> On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: >> >> >> >>> I have Freeswitch setup and working as a single tenant >>> system mostly using the default configuration. Trying to >>> convert to a multitenant environment, I have used both the >>> Multi-tenant and Multiple Companies wiki's. I get the phone >>> to register, can call out using the external profile to a >>> ITSP, can call music on hold; however I can not call other >>> users in the company. >>> It appears that when logged in with single company and >>> default context it sucessfully calls other internal phones >>> with bridge to >>> "sofia/internal/sip:extersion at public-IP:translated-port"; >>> however when I log into "Company1" with the phones, it tries >>> "sofia/internal/dialed-extension at Company1" ... I also get >>> "User not Registered". The dialplans are the same either >>> way. >>> >>> Any ideas? >>> >>> Thanks >>> John >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From lcm at marshap.com Tue Dec 22 10:50:04 2009 From: lcm at marshap.com (Larry Marshall) Date: Tue, 22 Dec 2009 10:50:04 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests Message-ID: <005101ca8337$9335ffb0$b9a1ff10$@com> I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml "external_sip_ip=stun:stun.freeswitch.org" On the original box in vars.xml "external_sip_port=5090" but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/be50aadd/attachment-0002.html From msc at freeswitch.org Tue Dec 22 11:14:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 11:14:36 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <005101ca8337$9335ffb0$b9a1ff10$@com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> Message-ID: <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall wrote: > I have set up a second FreeSWITCH box on the same LAN. I have v16018 > installed on it and have changed nothing. > > > > I configured a Polycom phone to register one of its four lines to this > second box, but it does not register. When looking at the console, there is > no activity. However, there is SIP activity on the box which I have captured > via ngrep. It looks like the phone is sending out REGISTER requests but > there is no response. The request on the pastebin repeats forever, with only > the timestamp varying. > > On the new box do "sofia status" - does the internal profile exist? > > > Is the problem that there are two FreeSWITCHes? Any suggestions on how I > can make it work? > > > > On the original and the new box in vars.xml "external_sip_ip=stun: > stun.freeswitch.org" > > On the original box in vars.xml "external_sip_port=5090" but in the new it > is 5080. > > > > Do I need to hardcode the external_sip_ip addresses in both boxes? > > > > http://pastebin.freeswitch.org/11600 > > > > Thanks Lars > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/641d2a22/attachment-0002.html From codecomplete at free.fr Tue Dec 22 11:24:55 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 22 Dec 2009 11:24:55 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all Message-ID: <26892767.post@talk.nabble.com> Hello I'm running "1.0.trunk (15841)" on Linux CentOS with a the default settings. After succesfully connecting a Windows PC running XLite (EyeBeam, really) and a GrandStream IP phone to Freeswitch, I try to make calls, but am having the following issues: 1. When calling XLite from GS, XLite rings, but when I pick up the call, the caller is sent to voice-mail right away ("the person on extension 1001 is not available") 2. When calling GS from XLite, the GS phone doesn't even ring. FWIW, the phones seem to have registered OK: freeswitch at internal> sofia status profile internal Registrations: ======================================================== Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. User: 1001 at 192.168.0.7 Contact: "Freeswitch" Agent: eyeBeam release 1104a stamp 54437 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) Host: centos.workgroup IP: 192.168.0.1 Port: 41380 Auth-User: 1001 Auth-Realm: 192.168.0.7 MWI-Account: 1001 at 192.168.0.7 Call-ID: 3f6d4ebebd5e829f at 192.168.0.9 User: 1003 at 192.168.0.7 Contact: "user" Agent: Grandstream BT120 1.1.0.3 Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) Host: centos.workgroup IP: 192.168.0.9 Port: 5060 Auth-User: 1003 Auth-Realm: 192.168.0.7 MWI-Account: 1003 at 192.168.0.7 ======================================================== Has someone seen this type of behavior? Thanks for any hint. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26892767.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Tue Dec 22 11:30:54 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 22 Dec 2009 11:30:54 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: <26893059.post@talk.nabble.com> I found the cause for #2: The GS phone was still configured to use NAT, even though both XLite and GS are located in the same, private LAN. Unchecking this on the GS phone solved the issue. But I'm still having issue #1, regardless of which phone is calling or being called: When the phone answers the call, I'm sent automatically to voice-mail. Could it be codec-related, or something like that? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yehavi.bourvine at gmail.com Tue Dec 22 11:35:55 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:35:55 +0200 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: Try tracing the calls from both sides with TCPDUMP or enable siptrace on FreeSwitch. I guess this will give you some clue. __Yehavi: 2009/12/22 Fred-145 > > Hello > > I'm running "1.0.trunk (15841)" on Linux CentOS with a the default > settings. > After succesfully connecting a Windows PC running XLite (EyeBeam, really) > and a GrandStream IP phone to Freeswitch, I try to make calls, but am > having > the following issues: > > 1. When calling XLite from GS, XLite rings, but when I pick up the call, > the > caller is sent to voice-mail right away ("the person on extension 1001 is > not available") > 2. When calling GS from XLite, the GS phone doesn't even ring. > > FWIW, the phones seem to have registered OK: > > freeswitch at internal> sofia status profile internal > Registrations: > ======================================================== > Call-ID: Yzc2MzFiMjVhNGQwNjE5YWU1OGZjNGMxMTg0NDIwNDA. > User: 1001 at 192.168.0.7 > Contact: "Freeswitch" > > Agent: eyeBeam release 1104a stamp 54437 > Status: Registered(UDP)(unknown) EXP(2008-01-01 03:34:00) > Host: centos.workgroup > IP: 192.168.0.1 > Port: 41380 > Auth-User: 1001 > Auth-Realm: 192.168.0.7 > MWI-Account: 1001 at 192.168.0.7 > > Call-ID: 3f6d4ebebd5e829f at 192.168.0.9 > User: 1003 at 192.168.0.7 > Contact: "user" > > Agent: Grandstream BT120 1.1.0.3 > Status: Registered(UDP)(unknown) EXP(2008-01-01 03:44:02) > Host: centos.workgroup > IP: 192.168.0.9 > Port: 5060 > Auth-User: 1003 > Auth-Realm: 192.168.0.7 > MWI-Account: 1003 at 192.168.0.7 > ======================================================== > > Has someone seen this type of behavior? > > Thanks for any hint. > -- > View this message in context: > http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26892767.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/f7721fb2/attachment-0002.html From yehavi.bourvine at gmail.com Tue Dec 22 11:39:15 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:39:15 +0200 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26893059.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <26893059.post@talk.nabble.com> Message-ID: It is usually CODEC related. probably the SIP messages has the cause inside. __Yehavi: 2009/12/22 Fred-145 > > I found the cause for #2: The GS phone was still configured to use NAT, > even > though both XLite and GS are located in the same, private LAN. Unchecking > this on the GS phone solved the issue. > > But I'm still having issue #1, regardless of which phone is calling or > being > called: When the phone answers the call, I'm sent automatically to > voice-mail. Could it be codec-related, or something like that? > > Thank you. > -- > View this message in context: > http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26893059.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/1f9cbe0e/attachment-0002.html From yehavi.bourvine at gmail.com Tue Dec 22 11:43:05 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Dec 2009 21:43:05 +0200 Subject: [Freeswitch-users] How to debug TLS handshake errors? In-Reply-To: <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> References: <2d9149cd0912170804v61d1aabeueb8e25aadf901c11@mail.gmail.com> <13CEA315-5ED6-4715-B5C2-7F68CECEF126@freeswitch.org> <45756176-AC3F-4E92-8560-DBDD8E8CEFC4@freeswitch.org> Message-ID: My distro is fedora 10 with all the current patches. SSLwatch fails to build and it seems more than a trivial change to make it work; however, it seems that the error message from Freeswitch tells it all... Is there any special debug statement in Freeswitch to see more about its TLS negotations? Thanks, __Yehavi: 2009/12/21 Brian West > You have to watch it with TLS. Make sure your distro didn't mess up your > SSL libs due to the recent vulnerability found. I havn't tested with my > polycom in a few weeks but it was working on my Polycom after I uploaded the > ca cert and marked it as trusted/used on the phone. > > /b > > On Dec 20, 2009, at 8:26 AM, Yehavi Bourvine wrote: > > > I am trying now to set a Polycom to work with FreeSwitch and TLS. I have > a Polycom-501 which does not have an internal certificate, thus only one-way > certificate validation is needed. I've downloaded the root certificate to he > Polyciom, and Freeswitch gives me the following error: > > > > Peer did not provide X.509 Certificate > > I understand that it tries to do mutual authentication which is not > possible in this case. How can I tell FreeSwitch to ignore the client's > certificate? > > > > BTW, I am running 1.0.5pre9, and it works ok using TLS with SNOM and > Yealink. > > > > Thanks! __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/357b25a9/attachment-0002.html From msc at freeswitch.org Tue Dec 22 11:44:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 11:44:14 -0800 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: References: <26892767.post@talk.nabble.com> Message-ID: <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine wrote: > Try tracing the calls from both sides with TCPDUMP or enable siptrace on > FreeSwitch. I guess this will give you some clue. > > __Yehavi: > Additionally, turn on debugging on the console and capture that output. If you use fs_cli it has debug output turned on by default. Pastebin that output and post the link in this thread. If you happen to look at the traces and figure it out then please let us know. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/9eaa9a83/attachment-0002.html From larclap at yahoo.com Tue Dec 22 11:46:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 22 Dec 2009 11:46:11 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> Message-ID: <006c01ca833f$6a43d890$3ecb89b0$@com> Yes, the internal profile exists. Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 192.168.10.25 alias internal ALIASED ============================================================================ ===================== 3 profiles 1 alias From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 22, 2009 11:15 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall wrote: I have set up a second FreeSWITCH box on the same LAN. I have v16018 installed on it and have changed nothing. I configured a Polycom phone to register one of its four lines to this second box, but it does not register. When looking at the console, there is no activity. However, there is SIP activity on the box which I have captured via ngrep. It looks like the phone is sending out REGISTER requests but there is no response. The request on the pastebin repeats forever, with only the timestamp varying. On the new box do "sofia status" - does the internal profile exist? Is the problem that there are two FreeSWITCHes? Any suggestions on how I can make it work? On the original and the new box in vars.xml "external_sip_ip=stun:stun.freeswitch.org" On the original box in vars.xml "external_sip_port=5090" but in the new it is 5080. Do I need to hardcode the external_sip_ip addresses in both boxes? http://pastebin.freeswitch.org/11600 Thanks Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/c351afd4/attachment-0002.html From mastermind202 at gmail.com Tue Dec 22 11:51:25 2009 From: mastermind202 at gmail.com (mm_202) Date: Tue, 22 Dec 2009 14:51:25 -0500 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 In-Reply-To: <200912171305.57498.yivzhenko@mksat.net> References: <200912171305.57498.yivzhenko@mksat.net> Message-ID: <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> Yuriy, The FS wiki has examples of how to control the BLF/MWI using events. I had no problem getting to work with my GXP2020. Let me know if you want some direct code examples. -- MM. On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko wrote: > Hallo All! > I need information about setup BLF on GXP2010/2020 phones with Freeswitch. > I search in Freeswitch Wiki and maillist archives but find no usable > information. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From a.alalousi at gmail.com Tue Dec 22 12:24:43 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 22 Dec 2009 20:24:43 +0000 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <4B2FAA29.4010405@aastral.net> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> Message-ID: Excellent work and answers. Thanks gentlemen. I'm firing off a new thread re: codecs et. al. Have a great Christmas and a wonderful, prosperous New Year. Regards, Ahmed. 2009/12/21 Bill W > I recently added an overview to this wiki page to help make things more > clear as to which ACL you need for different purposes. > > http://wiki.freeswitch.org/wiki/ACL#Overview > > Thanks, > Bill W. > > > Mathieu Rene wrote: > > Check out: http://wiki.freeswitch.org/wiki/ACL#Users > > > > It'll automatically add users with a cidr= attribute to the ACL list. > > This way you can set channel variables in the users and use them through > > your dialplan, all authenticated by ip address. > > > > Cheers, > > > > Mathieu Rene > > Avant-Garde Solutions Inc > > Office: + 1 (514) 664-1044 x100 > > Cell: +1 (514) 664-1044 x200 > > mrene at avgs.ca > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/53ba0867/attachment-0002.html From a.alalousi at gmail.com Tue Dec 22 12:55:41 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Tue, 22 Dec 2009 20:55:41 +0000 Subject: [Freeswitch-users] Codecs and things Message-ID: Hello people, Can someone please clear the following ambiguities with codecs: 1. Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki is not clear in this regard 2. When an A-leg has negotiated a pass-through media codec, can the B-leg be transcoded into a non-pass-through codec, and vice-versa ? think A-leg incoming with a G.729 codec, and target for B-leg needs to be setup with a GSM-codec, say 3. Where in the developer's set of documentation are codecs discussed ? I would like to start porting some code of mine for G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 support is proving quite a hindrance, and there is no clear direction from the dev community as to when the same will be available. Incidentally, any news on this effort ? where are we with code, and what's an ETA for a Beta ? 4. On the same lines as (3) above, there is a codec dev template in the source tree. Again, where can I find documentation relating to this ? the template has hardly any docs at all. Best regards and warm wishes for a Merry Christmas and a great New Year to one and all. Ahmed. -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/4f015cf1/attachment-0002.html From msc at freeswitch.org Tue Dec 22 14:17:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 14:17:23 -0800 Subject: [Freeswitch-users] FreeSWITCH 1.0.5pre10 is now available Message-ID: <87f2f3b90912221417l3b4b7f00pe7a4c7775ce2d85a@mail.gmail.com> It's upgrade-and-test time! The new release announcement is on the main FreeSWITCH page: http://www.freeswitch.org/node/224 Please update, test, and report back bugs and questions. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/118432aa/attachment-0002.html From msc at freeswitch.org Tue Dec 22 14:20:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Dec 2009 14:20:32 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <006c01ca833f$6a43d890$3ecb89b0$@com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> Message-ID: <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: > Yes, the internal profile exists. > > > > Name Type > Data State > > > ================================================================================================= > > internal profile sip:mod_sofia at 192.168.10.25:5060 > RUNNING (0) > > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > > external profile sip:mod_sofia at 192.168.10.25:5080 > RUNNING (0) > > example.com gateway sip:joeuser at example.com > NOREG > > 192.168.10.25 alias > internal ALIASED > > > ================================================================================================= > > 3 profiles 1 alias > > > I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/b126118d/attachment-0002.html From vinuth.madinur at gmail.com Tue Dec 22 14:28:03 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 03:58:03 +0530 Subject: [Freeswitch-users] Choosing a Codec. Message-ID: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> Hi, I am playing a file to a landline number. the format of the file is as follows: [root at static-host var]# file message.wav message.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz In my vars.xml file I have used the following codec prefs: However, when freeswitch plays it, it always chooses the L16 at 8000hz codec. I'm not understanding why this is so. EXECUTE sofia/external/5135692990 at 208.78.161.197 playback(/var/message.wav) 2009-12-22 17:16:57.357048 [DEBUG] switch_ivr_play_say.c:1135 Codec Activated L16 at 8000hz 1 channels 20ms 2009-12-22 17:17:30.777182 [DEBUG] switch_ivr_play_say.c:1429 done playing file My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. Have I configured it wrong or does this transcoding always happen? Thanks, Vinuth. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/3d272e5d/attachment-0002.html From kristoff.bonne at skypro.be Tue Dec 22 14:06:47 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Tue, 22 Dec 2009 23:06:47 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch Message-ID: <4B3142F7.1080600@skypro.be> Hi all, This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" device for just 15 euro. This is a device which has on one side a USB-connector and on the other side 2 RJ-11 connectors (one FXO and one FSX). Internally, the device seams to contain a tigerjet 560C chipset. (see here: http://www.tjnet.com/chips/tiger560C.htm) What is interesting on this device is that is uses standard USB device-classes that are by default supported by most operating-systems: usb-sound and usb-hid. When I connect it to my server (mac mini 3G running debian), the system automatically recognises these two classes [168391.922479] usbcore: registered new interface driver hiddev [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on usb-0001:10:1b.1-1 [168391.939548] usbcore: registered new interface driver usbhid [168391.943984] usbhid: v2.6:USB HID core driver [168392.154596] usbcore: registered new interface driver snd-usb-audio And -behold- when I connect a handset in one of the port, I even get a dialtone and I can sent out DTMF-dialtone which are somehow partly (But I have no idea what program actually generates this dialtone !!!) Now, the question: Any idea if / how this can incorperated into freeswitch? Is there a way to use this device to connect a phone to freeswitch without having to go throu a SIP-client first. Cheerio! Kr. Bonne. -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/498362d4/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/498362d4/attachment-0002.bin From freeswitch at aastral.net Tue Dec 22 14:38:10 2009 From: freeswitch at aastral.net (Bill W) Date: Tue, 22 Dec 2009 17:38:10 -0500 Subject: [Freeswitch-users] Authenticating end points by IP In-Reply-To: <007701ca828f$f414f9b0$dc3eed10$@com> References: <3B4C5E27-5B70-42E5-B179-FAF1938DA2A4@avgs.ca> <4B2FAA29.4010405@aastral.net> <007701ca828f$f414f9b0$dc3eed10$@com> Message-ID: <4B314A52.8080808@aastral.net> Hello Lars, You can apply any acl to any profile. What you should do really depends on what you want to accomplish. But let's take a simple example. Let's say you want to allow any phone on your internal network (192.168.0.0/24) to connect to your internal profile and make calls without having to provide a password. Then you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from. For your external profile, you may not want to have any ACLs at all, as you may not want to limit which IPs can connect to your switch to send you incoming calls. BUT, you need to make sure the dialplan connected to that external profile doesn't allow anyone to dial numbers that are not hosted on your system without proper authentication or controls. And believe me, people WILL try to do that. I've set up my system to email me whenever this happens and I have logged over 100 attempts to dial international numbers just since December 3rd. Hope this helps, Bill Lars Zeb wrote: > Bill, > > Thanks for your ACL Overview. Perhaps you can help me understand more > clearly. > > If you include the "local-network-acl" and "apply-inbound-acl" params in the > sip_profiles and setup the list for "localnet.auto" in acl.conf.xml, does > this mean you do not have to include the cidr attribute for individual > extensions in the directory/default folder? > > Is "apply-inbound-acl" supposed to exist in both internal and external > profiles while "apply-inbound-acl" is only in the internal? > > Thanks, Lars > From brian at freeswitch.org Tue Dec 22 14:39:06 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 16:39:06 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> Message-ID: <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> Why? You don't have to avoid it... why bother? /b On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: > My basic intent is to avoid on-the-fly transcoding, while having a high quality audio playing on PSTN. From vinuth.madinur at gmail.com Tue Dec 22 14:54:47 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 04:24:47 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> Message-ID: <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:09 AM, Brian West wrote: > Why? You don't have to avoid it... why bother? > > /b > > On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote: > > > My basic intent is to avoid on-the-fly transcoding, while having a high > quality audio playing on PSTN. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/20e9b85f/attachment-0002.html From brian at freeswitch.org Tue Dec 22 15:11:16 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Dec 2009 17:11:16 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> Message-ID: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. Can you elaborate on your setup a bit more? /b On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. > > Thanks, > Vinuth. From Mailings at kh-dev.de Tue Dec 22 15:22:30 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Wed, 23 Dec 2009 00:22:30 +0100 Subject: [Freeswitch-users] Make error... Message-ID: Hi all, I just downloaded the newest trunk about 5 minutes ago and I got the following make error on Ubuntu 8.04: gcc -E /usr/src/freeswitch/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive mkdir .libs Compiling src/switch_apr.c ... cc1: warnings being treated as errors src/switch_apr.c: In function 'switch_uuid_parse': src/switch_apr.c:899: warning: control reaches end of non-void function make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make: *** [all] Error 2 Regards, Klaus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/c47eed65/attachment-0002.html From jason at jasonjgw.net Tue Dec 22 15:38:29 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Dec 2009 10:38:29 +1100 Subject: [Freeswitch-users] Make error... In-Reply-To: References: Message-ID: <20091222233829.GA8702@jdc.jasonjgw.net> Klaus Hochlehnert wrote: > src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. From Mailings at kh-dev.de Tue Dec 22 16:20:01 2009 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Wed, 23 Dec 2009 01:20:01 +0100 Subject: [Freeswitch-users] Make error... In-Reply-To: <20091222233829.GA8702@jdc.jasonjgw.net> References: <20091222233829.GA8702@jdc.jasonjgw.net> Message-ID: I was on 16031. Now I downloaded 16032 and currently the make is running. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, December 23, 2009 12:38 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Make error... Klaus Hochlehnert wrote: > src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From rupa at rupa.com Tue Dec 22 16:45:37 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 22 Dec 2009 18:45:37 -0600 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji wrote: > Hello people, > > Can someone please clear the following ambiguities with codecs: > > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy > Media mode, or does FS need to be running in bypass-media ? the Wiki is not > clear in this regard Yes, you can use proxy media, bypass media, or even regular mode if you don't transcode (special for g729). Proxy media is really a special hack that should only be used for T38 passthrough. If you are using it for other purposes, think about it some more.... > When an A-leg has negotiated a pass-through media codec, can the B-leg be > transcoded into a non-pass-through codec, and vice-versa ? think A-leg > incoming with a G.729 codec, and target for B-leg needs to be setup with a > GSM-codec, say That would require transcoding - which can't be done if the codec is pass-through. > Where in the developer's set of documentation are codecs discussed ? I would > like to start porting some code of mine for G.729a/b/ab form a ti DSP > platform to FS. FS lacking full G.729 support is proving quite a hindrance, > and there is no clear direction from the dev community as to when the same > will be available. Incidentally, any news on this effort ? where are we with > code, and what's an ETA for a Beta ? I'd say look at the broadvoice or other simple self-contained codecs are done. Currently the only supported g729 solution is to use a digium board with mod_dahdi_codec. I don't have any info on a software based g729 solution. > On the same lines as (3) above, there is a codec dev template in the source > tree. Again, where can I find documentation relating to this ? the template > has hardly any docs at all. > > Best regards and warm wishes for a Merry Christmas and a great New Year to > one and all. > > Ahmed. > > > -- > Ahmed A. Ibrahim-Naji Al-Alousi > Ph.D., MIEE, MBCS > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From rupa at rupa.com Tue Dec 22 16:47:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 22 Dec 2009 18:47:52 -0600 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B3142F7.1080600@skypro.be> References: <4B3142F7.1080600@skypro.be> Message-ID: Interesting. It would have to do more than just dialtone/dtmf though. Need call control, caller id, etc. What do they ship with it as far as drivers go? On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne wrote: > Hi all, > > > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) > > > What is interesting on this device is that is uses standard USB > device-classes that are by default supported by most operating-systems: > usb-sound and usb-hid. > > > When I connect it to my server (mac mini 3G running debian), the system > automatically recognises these two classes > > [168391.922479] usbcore: registered new interface driver hiddev > [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on > usb-0001:10:1b.1-1 > [168391.939548] usbcore: registered new interface driver usbhid > [168391.943984] usbhid: v2.6:USB HID core driver > [168392.154596] usbcore: registered new interface driver snd-usb-audio > > > And -behold- when I connect a handset in one of the port, I even get a > dialtone and I can sent out DTMF-dialtone which are somehow partly > (But I have no idea what program actually generates this dialtone !!!) > > > > Now, the question: > Any idea if / how this can incorperated into freeswitch? Is there a way > to use this device to connect a phone to freeswitch without having to go > throu a SIP-client first. > > > > Cheerio! Kr. Bonne. > > -- > jabber/gtalk: kristoff at krbonne.net > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From dave at 3c.co.uk Tue Dec 22 19:29:28 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 23 Dec 2009 03:29:28 +0000 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> Message-ID: <1261538968.18473.55.camel@local.freepabx.com> On the other hand, a u-law WAV turned into L16 and then back to u-law to be sent down the line shouldn't suffer any alteration at all - if it does, the there's something wrong with the translation. The quality dropping over time is almost certainly down to something else. Vinuth -can you get a recording to compare with the original? --Dave > If its degrading like that you have bigger issues... the sound files played from wav files vs raw PCM files is NO different on a land line and I speak from very many years of experience... your wav files are ulaw in wav containers thus will never play native which might just be part of your problem. You would have to have raw headerless data in a .PCMU file for it to play native. > > Can you elaborate on your setup a bit more? > > /b > > On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > > > The audio quality is a lot different when it plays on the landline. And the quality degrades a bit when the message played is lengthy >30s. So I thought it would be better if I have the file in mu-law and play it as is.. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Tue Dec 22 19:44:18 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 22 Dec 2009 22:44:18 -0500 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <9dc4a1670912221944y3c302d37gc466f45a5d60df0f@mail.gmail.com> Have you considered GIPS http://www.gipscorp.com/products/overview.php ? -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/99a38e79/attachment-0002.html From JCasale at activenetwerx.com Tue Dec 22 19:58:30 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 23 Dec 2009 03:58:30 +0000 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: >> Am I correct in presuming that Freeswitch will answer a fax from a local zap based user >> just like it does from an FXO port connected to a POTS line? What I hope to do here is >> catch any call made from that extension (the zap based fax machine/user) and push its >> call into the fax module. > > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it gets dialtone > and dials. Whatever it dials is put into ${destination_number} just like any SIP phone that > dials. This extension looks ok. Try it out and let us know how it goes. > -MC Michael, It worked well, there was however a humorous moment: I was testing with my own shell script that simply emailed me directly to my postfix gateway, my exchange server and mua understood the uuencoded attachment so once it started working I modified the script to send to our fax service. Well they didn't understand uuencode so the attachment, a single page tiff, got faxed as 23 pages of binary :) I used mutt with a redirection to a specific muttrc which understands mime encoding which should work everywhere... Thanks for the help, you've made an office full of people happy... jlc From mike at jerris.com Tue Dec 22 20:13:21 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:13:21 -0500 Subject: [Freeswitch-users] WARNING On Inbound Call Question In-Reply-To: <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> References: <191c3a030911031423y58905d44mc00a7d949573ff86@mail.gmail.com> <1EA0C7D75E6E434AAC6E7D1273752004@greyhawk.tonecommander.com> <0A42096F-7F6E-4CDE-BB6C-2817A54E8228@freeswitch.org> Message-ID: <4D7D2B1F-7880-4DDE-8F71-1D6F78B6E212@jerris.com> If this is using prid it also requires the latest drivers from sangoma. I am pretty sure these are just in dev snapshots not release drivers yet. Something 3.5.8.6 or later iirc. Mike On Dec 21, 2009, at 7:52 PM, Brian West wrote: > You know that warning is meaningless. Search the archives we have > talked about this to no end it seems. > > And I'm sure Moy fixed this. > > /b > > On Dec 21, 2009, at 6:24 PM, Jerry Richards wrote: > >> Okay, I upgraded to 1.0.5pre9 and tried this test again and I do >> not see the WARNING in the Freeswitch log. However, it still >> behaves the same way. That is, the internal callee rings for about >> 12 seconds, then stops ringing, and the PSTN caller just hears >> ringback for about 60 seconds and is not given the opportunity to >> leave voice mail. In contrast, an internal-to-internal call will >> go to voice mail after 30 seconds. >> >> I put a new 11595 log into the pastebin. Is there some Sangoma >> Wanpipe driver (or Freeswitch) setting that would correct this? >> >> Best Regards, >> Jerry >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/2597ed69/attachment-0002.html From mike at jerris.com Tue Dec 22 20:31:16 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:31:16 -0500 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: References: Message-ID: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> For the path in the dialplan I don't think we have any right now but file a bug on jira and I can try to add them. As for something in the script itself that is a bit more work but if anyone has a patch to inject some vars into scripts like that it would be a nice addition. Mike On Dec 21, 2009, at 7:03 PM, "Joseph L. Casale" wrote: > Searching through the wiki for any indication as to what if any > variables exist > > for the install location in that I can leverage in a script. > > > > Can anyone point me along, I can?t seem to find anything. I want to > place a shell > > script in /opt/freeswitch/scripts that needs a reference to a conf > file that a binary > > it runs is calling. > > > > So now I have in two places hardcoded paths that I was hoping to > avoid, in the dialplan > > and in the shell script. When either of these is run, does there > exist something like > > > > shell_script.sh"/> > > > > and the same for use inside the shell script? > > > > Thanks! > jlc > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:36:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:36:22 -0500 Subject: [Freeswitch-users] Force endpoint to use rfc2833 for dtmf In-Reply-To: <4B30B01B.30809@gmx.net> References: <4B30B01B.30809@gmx.net> Message-ID: Not sure if we have an option to disable info. Even without this, dtmf should go across the bridge fine. Please open up a bug on jira about this Mike On Dec 22, 2009, at 6:40 AM, Peter P GMX wrote: > Hello, > > in a bigger installation with some thousand endpoints in the field we > see, that the endpoint equipment is always using INFO messages > (standard > setting is auto, so the endpoint decides which method to use). I > have 2 > questions to that scenario: > > 1. Is there a way that Freeswitch forces/restricts the endpoint to > use rfc2833 or not to send to allow INFO in the invite message? > 2. Currently INFO messages do not get forwarded from the caller > through freeswitch to called endpoint. How can we enable that FS > is fowarding the INFO messages? > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:38:49 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:38:49 -0500 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> Message-ID: <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins wrote: > > > On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: > Yes, the internal profile exists. > > > > Name > Type Data State > > === > === > === > === > === > === > === > === > === > ====================================================================== > > internal profile > sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) > > internal-ipv6 profile sip:mod_sofia@[:: > 1]:5060 RUNNING (0) > > external profile > sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) > > example.com gateway > sip:joeuser at example.com NOREG > > 192.168.10.25 alias > internal ALIASED > > === > === > === > === > === > === > === > === > === > ====================================================================== > > 3 profiles 1 alias > > > > > I would do a sanity check at this point: put this box and one phone > on a completely separate network with nothing else and see what > happens. > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/31a1d355/attachment-0002.html From mike at jerris.com Tue Dec 22 20:48:36 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:48:36 -0500 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: We expect the g729 sometime very soon, weeks not months away. Mike On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote: > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji > wrote: >> Hello people, >> >> Can someone please clear the following ambiguities with codecs: >> >> Are we definitively able to run pass-through codecs (e.g. G.729) in >> Proxy >> Media mode, or does FS need to be running in bypass-media ? the >> Wiki is not >> clear in this regard > > Yes, you can use proxy media, bypass media, or even regular mode if > you don't transcode (special for g729). Proxy media is really a > special hack that should only be used for T38 passthrough. If you are > using it for other purposes, think about it some more.... > >> When an A-leg has negotiated a pass-through media codec, can the B- >> leg be >> transcoded into a non-pass-through codec, and vice-versa ? think A- >> leg >> incoming with a G.729 codec, and target for B-leg needs to be setup >> with a >> GSM-codec, say > > That would require transcoding - which can't be done if the codec is > pass-through. > >> Where in the developer's set of documentation are codecs >> discussed ? I would >> like to start porting some code of mine for G.729a/b/ab form a ti DSP >> platform to FS. FS lacking full G.729 support is proving quite a >> hindrance, >> and there is no clear direction from the dev community as to when >> the same >> will be available. Incidentally, any news on this effort ? where >> are we with >> code, and what's an ETA for a Beta ? > > I'd say look at the broadvoice or other simple self-contained codecs > are done. Currently the only supported g729 solution is to use a > digium board with mod_dahdi_codec. > > I don't have any info on a software based g729 solution. > >> On the same lines as (3) above, there is a codec dev template in >> the source >> tree. Again, where can I find documentation relating to this ? the >> template >> has hardly any docs at all. >> >> Best regards and warm wishes for a Merry Christmas and a great New >> Year to >> one and all. >> >> Ahmed. >> >> >> -- >> Ahmed A. Ibrahim-Naji Al-Alousi >> Ph.D., MIEE, MBCS >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Dec 22 20:52:00 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Dec 2009 23:52:00 -0500 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> That being said, ulaw l16 alaw will cause degredation and any other modifications such as volume adjustment in this path will make it worse. Tha being said that does not sound like what you are experiencing Mike On Dec 22, 2009, at 10:29 PM, David Knell wrote: > On the other hand, a u-law WAV turned into L16 and then back to u- > law to > be sent down the line shouldn't suffer any alteration at all - if it > does, the there's something wrong with the translation. > > The quality dropping over time is almost certainly down to something > else. Vinuth -can you get a recording to compare with the original? > > --Dave > > >> If its degrading like that you have bigger issues... the sound >> files played from wav files vs raw PCM files is NO different on a >> land line and I speak from very many years of experience... your >> wav files are ulaw in wav containers thus will never play native >> which might just be part of your problem. You would have to have >> raw headerless data in a .PCMU file for it to play native. >> >> Can you elaborate on your setup a bit more? >> >> /b >> >> On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: >> >>> The audio quality is a lot different when it plays on the >>> landline. And the quality degrades a bit when the message played >>> is lengthy >30s. So I thought it would be better if I have the >>> file in mu-law and play it as is.. >>> >>> Thanks, >>> Vinuth. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From JCasale at activenetwerx.com Tue Dec 22 21:03:15 2009 From: JCasale at activenetwerx.com (Joseph L. Casale) Date: Wed, 23 Dec 2009 05:03:15 +0000 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> References: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> Message-ID: >For the path in the dialplan I don't think we have any right now but >file a bug on jira and I can try to add them. As for something in the >script itself that is a bit more work but if anyone has a patch to >inject some vars into scripts like that it would be a nice addition. > >Mike Ok, signed up for an account, where does the dialplan part go, FSCORE? Thanks for the help! jlc From marc at kasteris.com Tue Dec 22 21:14:16 2009 From: marc at kasteris.com (Marc Orenberg) Date: Tue, 22 Dec 2009 21:14:16 -0800 (PST) Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking Message-ID: <707756.35973.qm@web50808.mail.re2.yahoo.com> Hello. I've written an application using mod_conference which often has two parties speaking at once and one party listening. When only one party is speaking, the sound quality is fine, but when a second party starts speaking while the first party is still speaking, the second party's voice is cut-off at the beginning, and both parties voices seem to get choppy, like maybe all of the packets aren't getting delivered properly. I'm experiencing this with the latest trunk version (16012). I have the "member-flags" variable set to "waste", and "comfort-noise" is set to "true". I'm not sure where the problem is coming from; I think if it was a VOIP issue I'd hear the same problem when only one party is speaking. Is there something in mod_conference which would try to filter out other voices when one voice is speaking? I'd really appreciate any suggestions about where to look to find this problem. Thanks in advance, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/0071969d/attachment-0002.html From mike at jerris.com Tue Dec 22 21:18:33 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 00:18:33 -0500 Subject: [Freeswitch-users] Variables for install directories In-Reply-To: References: <7BB7BEBD-9CB2-4FD8-B6A3-AAF075068649@jerris.com> Message-ID: <9C787E83-4C76-42F0-85D9-00BE7D80F916@jerris.com> Sounds right to me, just assign it to me if it lets you Mike On Dec 23, 2009, at 12:03 AM, "Joseph L. Casale" wrote: >> For the path in the dialplan I don't think we have any right now but >> file a bug on jira and I can try to add them. As for something in >> the >> script itself that is a bit more work but if anyone has a patch to >> inject some vars into scripts like that it would be a nice addition. >> >> Mike > > Ok, signed up for an account, where does the dialplan part go, FSCORE? > Thanks for the help! > jlc > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From rob4manhere at gmail.com Tue Dec 22 21:23:10 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Tue, 22 Dec 2009 23:23:10 -0600 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <707756.35973.qm@web50808.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> Message-ID: <07801A24-05AA-4E78-9128-04CA349F59E6@gmail.com> Try setting your energy-level down, at 0 for instance. If it helps, then increase until you find a happy medium. On Dec 22, 2009, at 11:14 PM, Marc Orenberg wrote: > Hello. I've written an application using mod_conference which often > has two parties speaking at once and one party listening. > When only one party is speaking, the sound quality is fine, but when > a second party starts speaking while the first party is still > speaking, the second party's > voice is cut-off at the beginning, and both parties voices seem to > get choppy, like maybe all of the packets aren't getting delivered > properly. > > I'm experiencing this with the latest trunk version (16012). > I have the "member-flags" variable set to "waste", and "comfort- > noise" is set to "true". > > I'm not sure where the problem is coming from; I think if it was a > VOIP issue I'd hear the same problem when only one party is speaking. > Is there something in mod_conference which would try to filter out > other voices when one voice is speaking? > > I'd really appreciate any suggestions about where to look to find > this problem. > > Thanks in advance, > Marc > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/42782894/attachment-0002.html From jason at jasonjgw.net Tue Dec 22 21:29:58 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Dec 2009 16:29:58 +1100 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <707756.35973.qm@web50808.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> Message-ID: <20091223052958.GA7600@jdc.jasonjgw.net> Marc Orenberg wrote: > Is there something in mod_conference which would try to filter out other > voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. From marc at kasteris.com Tue Dec 22 21:45:57 2009 From: marc at kasteris.com (Marc Orenberg) Date: Tue, 22 Dec 2009 21:45:57 -0800 (PST) Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <20091223052958.GA7600@jdc.jasonjgw.net> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> Message-ID: <568455.74607.qm@web50801.mail.re2.yahoo.com> Thanks Rob, thanks Jason. I'm going to try this first thing tomorrow. The "energy-level" paramter is described in the file as, "Energy level required for audio to be sent to the other users", so one would think that this would have no effect if member-flags is set to "waste", right? ________________________________ From: Jason White To: freeswitch-users at lists.freeswitch.org Sent: Wed, December 23, 2009 12:29:58 AM Subject: Re: [Freeswitch-users] mod_conference voice problems when two parties speaking Marc Orenberg wrote: > Is there something in mod_conference which would try to filter out other > voices when one voice is speaking? Try reducing the energy level parameter in case this is the issue. It's 7/8/9 on the key pad during the call, or via the conference command, or the settings in conference.conf.xml. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091222/7fd9939c/attachment-0002.html From dule.maillist at gmail.com Tue Dec 22 22:00:51 2009 From: dule.maillist at gmail.com (Dan Le) Date: Wed, 23 Dec 2009 01:00:51 -0500 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <568455.74607.qm@web50801.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> <568455.74607.qm@web50801.mail.re2.yahoo.com> Message-ID: <914fc92a0912222200y64963743n7a6f82d48cca3247@mail.gmail.com> No, from my understanding that's not how it works. Waste just means it'll always send RTP packets, doesn't mean it will contain audio... so if you have audio that's under your energy threshold, you still won't hear it. Dan On Wed, Dec 23, 2009 at 12:45 AM, Marc Orenberg wrote: > Thanks Rob, thanks Jason. > I'm going to try this first thing tomorrow. > The "energy-level" paramter is described in the file as, "Energy level > required for audio to be sent to the other users", so one would think that > this would have no effect if member-flags is set to "waste", right? > > ------------------------------ > *From:* Jason White > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Wed, December 23, 2009 12:29:58 AM > *Subject:* Re: [Freeswitch-users] mod_conference voice problems when two > parties speaking > > Marc Orenberg wrote: > > Is there something in mod_conference which would try to filter out other > > voices when one voice is speaking? > > Try reducing the energy level parameter in case this is the issue. It's > 7/8/9 > on the key pad during the call, or via the conference command, or the > settings > in conference.conf.xml. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/27a5c779/attachment-0002.html From rob4manhere at gmail.com Tue Dec 22 22:03:12 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 23 Dec 2009 00:03:12 -0600 Subject: [Freeswitch-users] mod_conference voice problems when two parties speaking In-Reply-To: <568455.74607.qm@web50801.mail.re2.yahoo.com> References: <707756.35973.qm@web50808.mail.re2.yahoo.com> <20091223052958.GA7600@jdc.jasonjgw.net> <568455.74607.qm@web50801.mail.re2.yahoo.com> Message-ID: <4B4B1517-6473-486A-B726-3454B0B68C3D@gmail.com> No, they are related by different. Waste means mod_conference is *sending* to the recipients all of the time.. whether the conference is silent or not. Energy-level is the hump each channels has to get over before mod_conference accepts audio from that line and includes that channel in the conference. By setting it to 0, it bridges all channels all of the time, whether they are quite or not. Rob On Dec 22, 2009, at 11:45 PM, Marc Orenberg wrote: > Thanks Rob, thanks Jason. > I'm going to try this first thing tomorrow. > The "energy-level" paramter is described in the file as, "Energy > level required for audio to be sent to the other users", so one > would think that this would have no effect if member-flags is set to > "waste", right? > > From: Jason White > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, December 23, 2009 12:29:58 AM > Subject: Re: [Freeswitch-users] mod_conference voice problems when > two parties speaking > > Marc Orenberg wrote: > > Is there something in mod_conference which would try to filter out > other > > voices when one voice is speaking? > > Try reducing the energy level parameter in case this is the issue. > It's 7/8/9 > on the key pad during the call, or via the conference command, or > the settings > in conference.conf.xml. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/bfc81026/attachment-0002.html From dave at 3c.co.uk Tue Dec 22 22:47:08 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 23 Dec 2009 06:47:08 +0000 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> Message-ID: <1261550828.18473.64.camel@local.freepabx.com> On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: > That being said, ulaw l16 alaw will cause degredation and any other > modifications such as volume adjustment in this path will make it > worse. Indeed. Storing prompts as 8k, 16-bit WAVs makes a lot of sense. [I am inordinately pleased with the above] --Dave From vinuth.madinur at gmail.com Tue Dec 22 23:08:53 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 12:38:53 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> Message-ID: <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> My setup is as follows: FreeSWITCH -> SIP Trunk -> PSTN. >From freeswitch, I'm making outbound calls using event socket via the "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is default settings. Using "playback" application, I'm playing a mu-law audio. I'm also starting the "vmd" application, so that I can replay the message on beep. Thanks for your suggestion on native format. I'll try it. Thanks, Vinuth. On Wed, Dec 23, 2009 at 4:41 AM, Brian West wrote: > If its degrading like that you have bigger issues... the sound files played > from wav files vs raw PCM files is NO different on a land line and I speak > from very many years of experience... your wav files are ulaw in wav > containers thus will never play native which might just be part of your > problem. You would have to have raw headerless data in a .PCMU file for it > to play native. > > Can you elaborate on your setup a bit more? > > /b > > On Dec 22, 2009, at 4:54 PM, Vinuth Madinur wrote: > > > The audio quality is a lot different when it plays on the landline. And > the quality degrades a bit when the message played is lengthy >30s. So I > thought it would be better if I have the file in mu-law and play it as is.. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9ee6022d/attachment-0002.html From vinuth.madinur at gmail.com Tue Dec 22 23:14:11 2009 From: vinuth.madinur at gmail.com (Vinuth Madinur) Date: Wed, 23 Dec 2009 12:44:11 +0530 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261550828.18473.64.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> <3B3C4AD8-BCB4-4F00-AFC2-6A3B1FF56576@jerris.com> <1261550828.18473.64.camel@local.freepabx.com> Message-ID: <910309030912222314hd5e6f81o7b0ca0c459d1d542@mail.gmail.com> On Wed, Dec 23, 2009 at 12:17 PM, David Knell wrote: > On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote: > > That being said, ulaw l16 alaw will cause degredation and any other > > modifications such as volume adjustment in this path will make it > > worse. > > Indeed. Storing prompts > as 8k, 16-bit WAVs > makes a lot of sense. > > [I am inordinately pleased with the above] > > --Dave > > Thanks, will try and get back. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9a4ae9f3/attachment-0002.html From kristoff.bonne at skypro.be Wed Dec 23 01:54:58 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Wed, 23 Dec 2009 10:54:58 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: References: <4B3142F7.1080600@skypro.be> Message-ID: <4B31E8F2.6040907@skypro.be> Hi Rupa, None. That's exactly the point. Everything has to be done over the usb "HID" interface. I've been reading about HID yesterday. HID is a usb interface that can be used for a large number of things, ranging from keyboard and game-controllers up to "water-cooling and PC-chassis" and point-of-sale or coin changer devices. It also has a telephony-interface: see page 69 to 72 of this document: http://www.usb.org/developers/devclass_docs/HID1_11.pdf This include call-control, on-hook/off-hook detection, DTMF-related things, etc. Now, the question is this: Is there a way to "plug" this all into freeswitch? Cheerio! Kr. Bonne. Rupa Schomaker schreef: > Interesting. It would have to do more than just dialtone/dtmf though. > Need call control, caller id, etc. What do they ship with it as far > as drivers go? > > On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne > wrote: > >> Hi all, >> >> >> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >> device for just 15 euro. This is a device which has on one side a >> USB-connector and on the other side 2 RJ-11 connectors (one FXO and one >> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >> (see here: http://www.tjnet.com/chips/tiger560C.htm) >> >> >> What is interesting on this device is that is uses standard USB >> device-classes that are by default supported by most operating-systems: >> usb-sound and usb-hid. >> >> >> When I connect it to my server (mac mini 3G running debian), the system >> automatically recognises these two classes >> >> [168391.922479] usbcore: registered new interface driver hiddev >> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on >> usb-0001:10:1b.1-1 >> [168391.939548] usbcore: registered new interface driver usbhid >> [168391.943984] usbhid: v2.6:USB HID core driver >> [168392.154596] usbcore: registered new interface driver snd-usb-audio >> >> >> And -behold- when I connect a handset in one of the port, I even get a >> dialtone and I can sent out DTMF-dialtone which are somehow partly >> (But I have no idea what program actually generates this dialtone !!!) >> >> >> >> Now, the question: >> Any idea if / how this can incorperated into freeswitch? Is there a way >> to use this device to connect a phone to freeswitch without having to go >> throu a SIP-client first. >> >> >> >> Cheerio! Kr. Bonne. >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/01e9921c/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/01e9921c/attachment-0002.bin From codecomplete at free.fr Wed Dec 23 02:59:47 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 02:59:47 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> Message-ID: <26897658.post@talk.nabble.com> mercutioviz wrote: > Additionally, turn on debugging on the console and capture that output. If > you use fs_cli it has debug output turned on by default. Thanks for the tip. I launched fs_cli, typed ""sofia profile internal siptrace on", and then made a call from XLite to the GS phone, with the same issue. I wish I could go through the debug messages in the CLI, but there's so much data that I can't even see the beginning. Is there a way to reduce the amount of information, eg. only displaying the SIP messages, or only displaying the lines that start with [debug] while ignoring those that start with [notice]? Thank you. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897658.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 23 03:26:56 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 03:26:56 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26897658.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> Message-ID: <26897904.post@talk.nabble.com> I guess I can limit the amount of debug data in the CLI by choosing the right debug level: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP What is the recommended way to debug SIP connections like I'm having? -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26897904.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Wed Dec 23 04:49:15 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 23 Dec 2009 20:49:15 +0800 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port Message-ID: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> Hi all, I'm using FS 1.5, doesn't somebody known something about this problem? My scenario is : A(FreeSwitch) B ------INVITE -------> <----100 Tring-------- <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 c=IN IP4 10.36.143.76 <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 -----200 OK ------> response for UPDATE message <---- 200 OK-------- response for INVITE message, with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 --------ACK ---------> The problem is, B changed the rtp port in UPDATE message and "200 OK" response message, but FS didn't do update , so it still send and receive data from port 55066. Is this a bug in FS? Does someone known something about this problem? Any advice is appreciated! -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ff823243/attachment-0002.html From codecomplete at free.fr Wed Dec 23 04:51:35 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 04:51:35 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26897904.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> <26897904.post@talk.nabble.com> Message-ID: <26901641.post@talk.nabble.com> BTW, here's the layout: http://img192.imageshack.us/img192/539/investigatingimmediater.jpg All hosts are located in the same LAN with network address 192.168.0.0/24 and are connected to the hub in the ADSL NAT router. Regardless of whether I use XLite, the GrandStream IP phone, or the analog handset connected to the Linksys 3102, I get the same error: The remote extension (target) rings, but when I pick up the call on the (target) phone, the call is terminated on the target end, and the source extension is immediately redirected to the target extension's voice-mail. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26901641.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From yivzhenko at mksat.net Wed Dec 23 05:30:24 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Wed, 23 Dec 2009 15:30:24 +0200 Subject: [Freeswitch-users] BLF on Grandstream GXP2020 In-Reply-To: <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> References: <200912171305.57498.yivzhenko@mksat.net> <63de75710912221151s339e7a64vcb1bb85589894c83@mail.gmail.com> Message-ID: <200912231530.25380.yivzhenko@mksat.net> Yes i'l be happy to see some working examples :) I can't fully understand how freeswitch conceptually manage presence events. And i not found any information about it in wiki. With default configuration fs sends some notifications to subscribed phones without use any external scripts, but this works very unpredictable. (in most situations BLF change state only once and only for dialing side. after reboot phone BLF state may be change again on some events or may not change any more) if i use the following script blf state changed but only once after reboot the phone #!/usr/bin/php addHeader("proto", "sip"); $e1->addHeader("from", "$Username@$Domain"); $e1->addHeader("login", "$Username@$Domain"); $e1->addHeader("event_type", "presence"); $e1->addHeader("alt_event_type", "dialog"); $e1->addHeader("Presence-Call-Direction", "outbound"); $e1->addHeader("answer-state", $State); $res = $esl1->sendEvent($e1); } SendEvent('220',"mydomain.net","confirmed"); sleep(4); SendEvent('220',"mydomain.net","terminated"); ?> On Tuesday 22 December 2009 21:51:25 mm_202 wrote: > Yuriy, > > The FS wiki has examples of how to control the BLF/MWI using events. > I had no problem getting to work with my GXP2020. > > Let me know if you want some direct code examples. > > -- MM. > > On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko wrote: > > Hallo All! > > I need information about setup BLF on GXP2010/2020 phones with > > Freeswitch. I search in Freeswitch Wiki and maillist archives but find no > > usable information. > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From a.alalousi at gmail.com Wed Dec 23 05:46:33 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Wed, 23 Dec 2009 13:46:33 +0000 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: Hi Rupa, Thanks for your feedback. I am currently running proxy mode, but whenever I try to force G.729 on in-bound and out-bound calls, I get an error in my logs to the effect the G.729 is only a pass-through codec. Both originator and reciepient have G.729 codecs. Have you seen this before ? Regards, Ahmed. 2009/12/23 Rupa Schomaker > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji wrote: > > Hello people, > > > > Can someone please clear the following ambiguities with codecs: > > > > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy > > Media mode, or does FS need to be running in bypass-media ? the Wiki is > not > > clear in this regard > > Yes, you can use proxy media, bypass media, or even regular mode if > you don't transcode (special for g729). Proxy media is really a > special hack that should only be used for T38 passthrough. If you are > using it for other purposes, think about it some more.... > > > When an A-leg has negotiated a pass-through media codec, can the B-leg be > > transcoded into a non-pass-through codec, and vice-versa ? think A-leg > > incoming with a G.729 codec, and target for B-leg needs to be setup with > a > > GSM-codec, say > > That would require transcoding - which can't be done if the codec is > pass-through. > > > Where in the developer's set of documentation are codecs discussed ? I > would > > like to start porting some code of mine for G.729a/b/ab form a ti DSP > > platform to FS. FS lacking full G.729 support is proving quite a > hindrance, > > and there is no clear direction from the dev community as to when the > same > > will be available. Incidentally, any news on this effort ? where are we > with > > code, and what's an ETA for a Beta ? > > I'd say look at the broadvoice or other simple self-contained codecs > are done. Currently the only supported g729 solution is to use a > digium board with mod_dahdi_codec. > > I don't have any info on a software based g729 solution. > > > On the same lines as (3) above, there is a codec dev template in the > source > > tree. Again, where can I find documentation relating to this ? the > template > > has hardly any docs at all. > > > > Best regards and warm wishes for a Merry Christmas and a great New Year > to > > one and all. > > > > Ahmed. > > > > > > -- > > Ahmed A. Ibrahim-Naji Al-Alousi > > Ph.D., MIEE, MBCS > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ddfd3f44/attachment-0002.html From a.alalousi at gmail.com Wed Dec 23 05:49:59 2009 From: a.alalousi at gmail.com (Ahmed Naji) Date: Wed, 23 Dec 2009 13:49:59 +0000 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: Hi Mike, Thanks for this. Is the alpha/dev version in the development repositories somewhere ? I would be very interested to participate. I have some code for the ti dsps which may or may not be useful. Also, I'd very much like to put in the effort to the ti DSPs within FS. Is this something that yourself and other devs would be interested in ? Regards, Ahmed. 2009/12/23 Michael Jerris > We expect the g729 sometime very soon, weeks not months away. > > Mike > > On Dec 22, 2009, at 7:45 PM, Rupa Schomaker wrote: > > > On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji > > wrote: > >> Hello people, > >> > >> Can someone please clear the following ambiguities with codecs: > >> > >> Are we definitively able to run pass-through codecs (e.g. G.729) in > >> Proxy > >> Media mode, or does FS need to be running in bypass-media ? the > >> Wiki is not > >> clear in this regard > > > > Yes, you can use proxy media, bypass media, or even regular mode if > > you don't transcode (special for g729). Proxy media is really a > > special hack that should only be used for T38 passthrough. If you are > > using it for other purposes, think about it some more.... > > > >> When an A-leg has negotiated a pass-through media codec, can the B- > >> leg be > >> transcoded into a non-pass-through codec, and vice-versa ? think A- > >> leg > >> incoming with a G.729 codec, and target for B-leg needs to be setup > >> with a > >> GSM-codec, say > > > > That would require transcoding - which can't be done if the codec is > > pass-through. > > > >> Where in the developer's set of documentation are codecs > >> discussed ? I would > >> like to start porting some code of mine for G.729a/b/ab form a ti DSP > >> platform to FS. FS lacking full G.729 support is proving quite a > >> hindrance, > >> and there is no clear direction from the dev community as to when > >> the same > >> will be available. Incidentally, any news on this effort ? where > >> are we with > >> code, and what's an ETA for a Beta ? > > > > I'd say look at the broadvoice or other simple self-contained codecs > > are done. Currently the only supported g729 solution is to use a > > digium board with mod_dahdi_codec. > > > > I don't have any info on a software based g729 solution. > > > >> On the same lines as (3) above, there is a codec dev template in > >> the source > >> tree. Again, where can I find documentation relating to this ? the > >> template > >> has hardly any docs at all. > >> > >> Best regards and warm wishes for a Merry Christmas and a great New > >> Year to > >> one and all. > >> > >> Ahmed. > >> > >> > >> -- > >> Ahmed A. Ibrahim-Naji Al-Alousi > >> Ph.D., MIEE, MBCS > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > >> users > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > -Rupa > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ahmed A. Ibrahim-Naji Al-Alousi Ph.D., MIEE, MBCS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/fdb0bf11/attachment-0002.html From steveu at coppice.org Wed Dec 23 05:55:00 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 23 Dec 2009 21:55:00 +0800 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <1261538968.18473.55.camel@local.freepabx.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <1261538968.18473.55.camel@local.freepabx.com> Message-ID: <4B322134.7030307@coppice.org> On 12/23/2009 11:29 AM, David Knell wrote: > On the other hand, a u-law WAV turned into L16 and then back to u-law to > be sent down the line shouldn't suffer any alteration at all - if it > does, the there's something wrong with the translation. > > The quality dropping over time is almost certainly down to something > else. Vinuth -can you get a recording to compare with the original? > A linear->ulaw->linear->ulaw->linear->ulaw chain *should* only loose quality on the first cycle. However: - Many ulaw implementation are buggy, because they are based on the same broken Sun code. People are gradually waking up and fixing this. The broken implementations can loose considerable quality on each cycle. - Anywhere people fiddle with gains, you will loose quality on each cycle. Steve From codecomplete at free.fr Wed Dec 23 06:22:28 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 06:22:28 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26901641.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <87f2f3b90912221144t2a30cd16xc9eb4267e17c7399@mail.gmail.com> <26897658.post@talk.nabble.com> <26897904.post@talk.nabble.com> <26901641.post@talk.nabble.com> Message-ID: <26902800.post@talk.nabble.com> FWIW, I downloaded and compiled the latest trunk (16041), and am still having this issue. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26902800.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From technical at ttnc.co.uk Wed Dec 23 06:26:40 2009 From: technical at ttnc.co.uk (TTNC - Technical) Date: Wed, 23 Dec 2009 14:26:40 +0000 Subject: [Freeswitch-users] RTP/RTCP media whilst recording Message-ID: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Hi There Our Freeswitch cluster receives inbound calls via a SIP trunk from our supplier. I currently have an issue where when a call is sent to voicemail using session:execute("record"), our supplier will terminate the call with a BYE approximately 30 seconds into the recording. They believe the reason for this is our Freeswitch servers are failing to send any RTP/RTCP media while in the recording stage, and therefor they think the call is dead. Is there a way to force Freeswitch to send RTP packets while in the recording stage that I'm missing? Oh, I'm running pretty much the latest svn truck. Any help appreciated. Thanks Russ From mike at jerris.com Wed Dec 23 06:29:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 09:29:38 -0500 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B31E8F2.6040907@skypro.be> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> Message-ID: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Of course there is a way. Depending on the interface your looking at either a freeswitch endpoiny module or an openzap module. Mike On Dec 23, 2009, at 4:54 AM, Kristoff Bonne wrote: > Hi Rupa, > > > None. That's exactly the point. > Everything has to be done over the usb "HID" interface. > > > I've been reading about HID yesterday. HID is a usb interface that > can be used for a large number of things, ranging from keyboard and > game-controllers up to "water-cooling and PC-chassis" and point-of- > sale or coin changer devices. > > > It also has a telephony-interface: > see page 69 to 72 of this document: http://www.usb.org/developers/devclass_docs/HID1_11.pdf > > This include call-control, on-hook/off-hook detection, DTMF-related > things, etc. > > > Now, the question is this: > Is there a way to "plug" this all into freeswitch? > > > > > Cheerio! Kr. Bonne. > > > Rupa Schomaker schreef: >> >> Interesting. It would have to do more than just dialtone/dtmf >> though. >> Need call control, caller id, etc. What do they ship with it as far >> as drivers go? >> >> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >> wrote: >> >>> Hi all, >>> >>> >>> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >>> device for just 15 euro. This is a device which has on one side a >>> USB-connector and on the other side 2 RJ-11 connectors (one FXO >>> and one >>> FSX). Internally, the device seams to contain a tigerjet 560C >>> chipset. >>> (see here: http://www.tjnet.com/chips/tiger560C.htm) >>> >>> >>> What is interesting on this device is that is uses standard USB >>> device-classes that are by default supported by most operating- >>> systems: >>> usb-sound and usb-hid. >>> >>> >>> When I connect it to my server (mac mini 3G running debian), the >>> system >>> automatically recognises these two classes >>> >>> [168391.922479] usbcore: registered new interface driver hiddev >>> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID >>> 06e6:c31c] on >>> usb-0001:10:1b.1-1 >>> [168391.939548] usbcore: registered new interface driver usbhid >>> [168391.943984] usbhid: v2.6:USB HID core driver >>> [168392.154596] usbcore: registered new interface driver snd-usb- >>> audio >>> >>> >>> And -behold- when I connect a handset in one of the port, I even >>> get a >>> dialtone and I can sent out DTMF-dialtone which are somehow partly >>> (But I have no idea what program actually generates this >>> dialtone !!!) >>> >>> >>> >>> Now, the question: >>> Any idea if / how this can incorperated into freeswitch? Is there >>> a way >>> to use this device to connect a phone to freeswitch without having >>> to go >>> throu a SIP-client first. >>> >>> >>> >>> Cheerio! Kr. Bonne. >>> >>> -- >>> jabber/gtalk: kristoff at krbonne.net >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >>> users >>> http://www.freeswitch.org >>> >>> >>> >> >> >> >> > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/cd3e0c5a/attachment-0002.html From steveu at coppice.org Wed Dec 23 06:31:30 2009 From: steveu at coppice.org (Steve Underwood) Date: Wed, 23 Dec 2009 22:31:30 +0800 Subject: [Freeswitch-users] Codecs and things In-Reply-To: References: Message-ID: <4B3229C2.4080109@coppice.org> On 12/23/2009 04:55 AM, Ahmed Naji wrote: > Hello people, > > Can someone please clear the following ambiguities with codecs: > > 1. Are we definitively able to run pass-through codecs (e.g. G.729) > in Proxy Media mode, or does FS need to be running in > bypass-media ? the Wiki is not clear in this regard > 2. When an A-leg has negotiated a pass-through media codec, can the > B-leg be transcoded into a non-pass-through codec, and > vice-versa ? think A-leg incoming with a G.729 codec, and target > for B-leg needs to be setup with a GSM-codec, say > 3. Where in the developer's set of documentation are codecs > discussed ? I would like to start porting some code of mine for > G.729a/b/ab form a ti DSP platform to FS. FS lacking full G.729 > support is proving quite a hindrance, and there is no clear > direction from the dev community as to when the same will be > available. Incidentally, any news on this effort ? where are we > with code, and what's an ETA for a Beta ? > 4. On the same lines as (3) above, there is a codec dev template in > the source tree. Again, where can I find documentation relating > to this ? the template has hardly any docs at all. > > Best regards and warm wishes for a Merry Christmas and a great New > Year to one and all. The G.729 codec for FS is in testing, and should be out so. If you really want to implement your own, TI DSP code is unlikely to be a good starting point. I assume that code is fixed point. You really need a floating point codec to get any decent speed on a PC. Pentiums and Athlons lack saturating arithmetic (MMX actually has partial saturating arithmetic, but it isn't much use for anything but image processing), so a fixed point implementation ends up very slow. Steve From mike at jerris.com Wed Dec 23 06:32:26 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Dec 2009 09:32:26 -0500 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port In-Reply-To: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> References: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> Message-ID: <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> There is no such thing as freeswitch 1.5. Have you tried latest svn trunk to see if this behavior is the same? Mike On Dec 23, 2009, at 7:49 AM, Lei Tang wrote: > Hi all, I'm using FS 1.5, doesn't somebody known something about > this problem? > My scenario is : > A(FreeSwitch) B > ------INVITE -------> > <----100 Tring-------- > <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 > c=IN IP4 10.36.143.76 > <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 > c=IN IP4 10.36.143.76 > -----200 OK ------> response for UPDATE message > <---- 200 OK-------- response for INVITE message, with > sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 > --------ACK ---------> > The problem is, B changed the rtp port in UPDATE message and "200 > OK" response message, but FS didn't do update , so it still send and > receive data from port 55066. > Is this a bug in FS? Does someone known something about this > problem? Any advice is appreciated! > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/6f87a4e2/attachment-0002.html From brian at freeswitch.org Wed Dec 23 06:55:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 08:55:12 -0600 Subject: [Freeswitch-users] RTP/RTCP media whilst recording In-Reply-To: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> References: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Message-ID: <481BF889-0446-409E-9A07-DEA61D30BAE2@freeswitch.org> What does pretty much mean to you? Can you give me an exact rev? /b On Dec 23, 2009, at 8:26 AM, TTNC - Technical wrote: > Oh, I'm running pretty much the latest svn truck. From lei.tlfly at gmail.com Wed Dec 23 06:56:14 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 23 Dec 2009 22:56:14 +0800 Subject: [Freeswitch-users] FS doesn't update rtp port when sip UPDATE message changed the rtp port In-Reply-To: <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> References: <50c41b4e0912230449r12ba2838p8226032dd3fc3b57@mail.gmail.com> <2980A888-685F-4E46-BBC6-488EFDAEF656@jerris.com> Message-ID: <50c41b4e0912230656p18400250yd5a85d5cb0981753@mail.gmail.com> Thanks Michael, sorry for my mistake, I'm using FS 1.0.5pre9, I'll try the lastest svn trunk. 2009/12/23 Michael Jerris > There is no such thing as freeswitch 1.5. Have you tried latest svn trunk > to see if this behavior is the same? > > Mike > > > On Dec 23, 2009, at 7:49 AM, Lei Tang wrote: > > Hi all, I'm using FS 1.5, doesn't somebody known something about this > problem? > My scenario is : > A(FreeSwitch) B > ------INVITE -------> > <----100 Tring-------- > <----180 Ring -------- with sdp m=audio 55066 RTP/AVP 0 120 c=IN > IP4 10.36.143.76 > <----UPDATE ------- with sdp m=audio 45486 RTP/AVP 0 120 c=IN IP4 > 10.36.143.76 > -----200 OK ------> response for UPDATE message > <---- 200 OK-------- response for INVITE message, with sdp > m=audio 45486 RTP/AVP 0 120 c=IN IP4 10.36.143.76 > --------ACK ---------> > The problem is, B changed the rtp port in UPDATE message and "200 OK" > response message, but FS didn't do update , so it still send and receive > data from port 55066. > Is this a bug in FS? Does someone known something about this problem? Any > advice is appreciated! > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/710a1f83/attachment-0002.html From brian at freeswitch.org Wed Dec 23 06:56:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 08:56:53 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> Message-ID: VMD will force a transcode anyway too. /b On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > My setup is as follows: > > FreeSWITCH -> SIP Trunk -> PSTN. > > From freeswitch, I'm making outbound calls using event socket via the "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is default settings. Using "playback" application, I'm playing a mu-law audio. I'm also starting the "vmd" application, so that I can replay the message on beep. > > Thanks for your suggestion on native format. I'll try it. > > Thanks, > Vinuth. From rupa at rupa.com Wed Dec 23 07:01:38 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Dec 2009 09:01:38 -0600 Subject: [Freeswitch-users] RTP/RTCP media whilst recording In-Reply-To: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> References: <5FE7DD79-C370-4821-96C0-24E92E8DD16D@ttnc.co.uk> Message-ID: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical wrote: > Hi There > > Our Freeswitch cluster receives inbound calls via a SIP trunk from our supplier. I currently have an issue where when a call is sent to voicemail using session:execute("record"), our supplier will terminate the call with a BYE approximately 30 seconds into the recording. > > They believe the reason for this is our Freeswitch servers are failing to send any RTP/RTCP media while in the recording stage, and therefor they think the call is dead. > > Is there a way to force Freeswitch to send RTP packets while in the recording stage that I'm missing? > > Oh, I'm running pretty much the latest svn truck. > > Any help appreciated. > > Thanks > > Russ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From anthony.minessale at gmail.com Wed Dec 23 07:22:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Dec 2009 09:22:26 -0600 Subject: [Freeswitch-users] Choosing a Codec. In-Reply-To: References: <910309030912221428x723f2918g1213325a54dfe1d9@mail.gmail.com> <796760D4-4C53-4773-82B4-E74F8EBFC10C@freeswitch.org> <910309030912221454s1cb05be1ja0b1c829f6f21d38@mail.gmail.com> <4FAE1A2F-EA4A-41CE-8E4E-46AD732B197D@freeswitch.org> <910309030912222308k3996aeb4qccb9cf0f9dc9a364@mail.gmail.com> Message-ID: <191c3a030912230722u7ab4c6bw70dd3e7170470ca2@mail.gmail.com> It's more than highly likely you have some other problem like jitter or a bad network connection. Not many people would be able to tell the difference between the sound of an 8k PCM file and the same file encoded to G711 just by listening to it unless there was a severe problem somewhere. Since you are behind NAT you are even more likely to experience drops etc. Record your files as 8k raw 16 bit PCM to get the best out of the file playback in FS and look elsewhere for your audio issues. You can always make sure you are using the latest build of FS to rule out any temporary issues in the code. On Wed, Dec 23, 2009 at 8:56 AM, Brian West wrote: > VMD will force a transcode anyway too. > > /b > > On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote: > > > My setup is as follows: > > > > FreeSWITCH -> SIP Trunk -> PSTN. > > > > From freeswitch, I'm making outbound calls using event socket via the > "external" profile. Except for the ext_rtp_ip and ext_sip_ip, everything is > default settings. Using "playback" application, I'm playing a mu-law audio. > I'm also starting the "vmd" application, so that I can replay the message on > beep. > > > > Thanks for your suggestion on native format. I'll try it. > > > > Thanks, > > Vinuth. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/5f0a921c/attachment-0002.html From codecomplete at free.fr Wed Dec 23 07:39:23 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Dec 2009 07:39:23 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26892767.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> Message-ID: <26903707.post@talk.nabble.com> More information: I can dial the default extensions like 9999 just fine. It's only when I call any of the IP phones (1001,1002,1003) that the call is immediately forwarded to the callee's voice-mail when the phone goes off the hook. To only keep the SIP messages in the fs_cli screen, typing "sofia loglevel all 0" followed by "sofia profile internal siptrace on" doesn't do the trick, so am unable to post the whole log yet. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26903707.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From john at acsol.net Wed Dec 23 07:43:17 2009 From: john at acsol.net (John) Date: Wed, 23 Dec 2009 08:43:17 -0700 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B30F0CD.8040703@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> Message-ID: <4B323A95.2070503@acsol.net> Still having this issue. Do separate domains need to be real fully qualified domains, or can they just be added as in Company1, 2, 3, etc? On 12/22/2009 9:16 AM, John wrote: > Thanks Brian. I did have both force-register-domain and > force-register-db-domain commented in both the internal.xml and > internal-ipv6.xml. The phones appear to register to the company1 domain, > as shown in sofia status profile company1; however I have noticed that > when I try to make a call to another a phone in the same domain, the > system is trying to call sofia/internal/1004 at company1 -- this is when we > get the message, user not registered. If I can the phones to just > register to the IP address of the machine, they call fine and is shows > sofia/internal/sip:1004 at phonesgatewayIPaddress. Is this a dialplan > problem? In both cases I am just using the sample dialplan. > > > > > On 12/22/2009 8:13 AM, Brian West wrote: > >> The force-register-domain and force-register-db-domain are set in the defaults so you can only do one domain. Remove those and you'll be able to do multiple domains. >> >> /b >> >> On Dec 21, 2009, at 6:15 PM, john at acsol.net wrote: >> >> >> >>> I have Freeswitch setup and working as a single tenant >>> system mostly using the default configuration. Trying to >>> convert to a multitenant environment, I have used both the >>> Multi-tenant and Multiple Companies wiki's. I get the phone >>> to register, can call out using the external profile to a >>> ITSP, can call music on hold; however I can not call other >>> users in the company. >>> It appears that when logged in with single company and >>> default context it sucessfully calls other internal phones >>> with bridge to >>> "sofia/internal/sip:extersion at public-IP:translated-port"; >>> however when I log into "Company1" with the phones, it tries >>> "sofia/internal/dialed-extension at Company1" ... I also get >>> "User not Registered". The dialplans are the same either >>> way. >>> >>> Any ideas? >>> >>> Thanks >>> John >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mattdfong at gmail.com Wed Dec 23 07:48:02 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 23 Dec 2009 07:48:02 -0800 Subject: [Freeswitch-users] forcing ptime settings Message-ID: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and it seemed to fix the really slow playback problem. but now I'm getting a [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 error and in some sessions the audio is playing back too fast (at 3x the speed). Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/2b56fcd2/attachment-0002.html From brian at freeswitch.org Wed Dec 23 07:48:48 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 09:48:48 -0600 Subject: [Freeswitch-users] Multitenant dialplans In-Reply-To: <4B323A95.2070503@acsol.net> References: <4b300f9a.313.2c10.1142196461@acsol.net> <2DD9D726-B0C3-4911-A4DC-AB2F03DE1E72@freeswitch.org> <4B30F0CD.8040703@acsol.net> <4B323A95.2070503@acsol.net> Message-ID: Yes DNS is required for this to work properly. /b On Dec 23, 2009, at 9:43 AM, John wrote: > Still having this issue. Do separate domains need to be real fully > qualified domains, or can they just be added as in Company1, 2, 3, etc? > From brian at freeswitch.org Wed Dec 23 07:55:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 09:55:58 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> Message-ID: <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> That usually means they are saying 30 but sending 10 which is broken.. you can't say hey i'm sending 30 and then send 10... find a new provider or beat them to death with a cluebat in hopes they fix their broken stuff. /b On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm having some trouble playing .wav files into the media stream using FreeSWITCH. > > The audio either comes out really slow, or really fast. So a 60 second .wav file is either finished playing in 90 seconds (really slow) or finishes playing in 20 seconds (really fast). I believe this is caused by different ptime values that are being setup in the session. In the FreeSWITCH console I often received this error > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 20 > > I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and it seemed to fix the really slow playback problem. > > but now I'm getting a > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant to say was 10 > > error and in some sessions the audio is playing back too fast (at 3x the speed). > > Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the ptime values? Are there any other work arounds? > > > --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/2346713c/attachment-0002.html From william.suffill at gmail.com Wed Dec 23 08:06:18 2009 From: william.suffill at gmail.com (William Suffill) Date: Wed, 23 Dec 2009 11:06:18 -0500 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Message-ID: <6b65470d0912230806j4a9c6297r8e393c5c735fe079@mail.gmail.com> I haven't played with any of these usb-to-rj11 in a long time but from what I do recall it is picked up as an audio device. That way a regular telephone can be used for mic/speaker for a softphone. Given that you should be able to get audio to/from it using portaudio for starters. Pretty basic and doesn't take advantage of all the features but it would be at least a place to look. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/98451cc2/attachment-0002.html From rupa at rupa.com Wed Dec 23 08:20:25 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Dec 2009 10:20:25 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> Message-ID: They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes. On Wed, Dec 23, 2009 at 9:55 AM, Brian West wrote: > That usually means they are saying 30 but sending 10 which is broken.. you > can't say hey i'm sending 30 and then send 10... find a new provider or beat > them to death with a cluebat in hopes they fix their broken stuff. > > /b > > On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > > I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm > having some trouble playing .wav files into the media stream using > FreeSWITCH. > > The audio either comes out really slow, or really fast. So a 60 second .wav > file is either finished playing in 90 seconds (really slow) or finishes > playing in 20 seconds (really fast). I believe this is caused by different > ptime values that are being setup in the session. In the FreeSWITCH console > I often received this error > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 20 > > I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and > it seemed to fix the really slow playback problem. > > but now I'm getting a > > [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they meant > to say was 10 > > error and in some sessions the audio is playing back too fast (at 3x the > speed). > > Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" the > ptime values? Are there any other work arounds? > > > --matt > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/e98ec9cc/attachment-0002.html From mattdfong at gmail.com Wed Dec 23 08:41:06 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 23 Dec 2009 08:41:06 -0800 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> Message-ID: <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> If I only care about outbound audio, is there a way to force the audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there still this same issue? --matt On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker wrote: > They don't operate their own voip gateways, just run an SBC in front of a > bunch of other providers. So someone they are reselling is using Sonus > gear. I use them to originate to some destinations but in the US I avoid > them due to the sonus stuff that pops up on certain routes. > > On Wed, Dec 23, 2009 at 9:55 AM, Brian West wrote: > >> That usually means they are saying 30 but sending 10 which is broken.. you >> can't say hey i'm sending 30 and then send 10... find a new provider or beat >> them to death with a cluebat in hopes they fix their broken stuff. >> >> /b >> >> On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: >> >> I use the SIP Termination service from ezcall inc (grnvoip.com) and I'm >> having some trouble playing .wav files into the media stream using >> FreeSWITCH. >> >> The audio either comes out really slow, or really fast. So a 60 second >> .wav file is either finished playing in 90 seconds (really slow) or finishes >> playing in 20 seconds (really fast). I believe this is caused by different >> ptime values that are being setup in the session. In the FreeSWITCH console >> I often received this error >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they >> meant to say was 20 >> >> I tried forcing the codec and ptime using absolute_codec_string='PCMU at 30i' and >> it seemed to fix the really slow playback problem. >> >> but now I'm getting a >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what they >> meant to say was 10 >> >> error and in some sessions the audio is playing back too fast (at 3x the >> speed). >> >> Is there a way I can force ptime to be 30 and avoid FreeSWITCH "fixing" >> the ptime values? Are there any other work arounds? >> >> >> --matt >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/7a595d8f/attachment-0002.html From mrene_lists at avgs.ca Wed Dec 23 08:53:25 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 23 Dec 2009 11:53:25 -0500 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> Message-ID: <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 23-Dec-09, at 11:41 AM, Matthew Fong wrote: > If I only care about outbound audio, is there a way to force the > audio packets FreeSWITCH sends to be of a certain ptime (like 30ms)? > Or is there still this same issue? > > --matt > > On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker wrote: > They don't operate their own voip gateways, just run an SBC in front > of a bunch of other providers. So someone they are reselling is > using Sonus gear. I use them to originate to some destinations but > in the US I avoid them due to the sonus stuff that pops up on > certain routes. > > On Wed, Dec 23, 2009 at 9:55 AM, Brian West > wrote: > That usually means they are saying 30 but sending 10 which is > broken.. you can't say hey i'm sending 30 and then send 10... find a > new provider or beat them to death with a cluebat in hopes they fix > their broken stuff. > > /b > > On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote: > >> I use the SIP Termination service from ezcall inc (grnvoip.com) and >> I'm having some trouble playing .wav files into the media stream >> using FreeSWITCH. >> >> The audio either comes out really slow, or really fast. So a 60 >> second .wav file is either finished playing in 90 seconds (really >> slow) or finishes playing in 20 seconds (really fast). I believe >> this is caused by different ptime values that are being setup in >> the session. In the FreeSWITCH console I often received this error >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what >> they meant to say was 20 >> >> I tried forcing the codec and ptime using >> absolute_codec_string='PCMU at 30i' and it seemed to fix the really >> slow playback problem. >> >> but now I'm getting a >> >> [WARNING] mod_sofia.c:808 We were told to use ptime 30 but what >> they meant to say was 10 >> >> error and in some sessions the audio is playing back too fast (at >> 3x the speed). >> >> Is there a way I can force ptime to be 30 and avoid FreeSWITCH >> "fixing" the ptime values? Are there any other work arounds? >> >> >> --matt > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/ca0ec922/attachment-0002.html From brian at freeswitch.org Wed Dec 23 08:57:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 10:57:26 -0600 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> Message-ID: You might also have to set the codec negotiation to scrooge /b On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote: > You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: > > > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/9d8e0313/attachment-0002.html From scott.torr.fs at letterboxes.org Wed Dec 23 10:00:43 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 05:00:43 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> Message-ID: <1261591243.5459.1351510407@webmail.messagingengine.com> Yes, I noticed the Jira for the situation where the where the fs controlled skype client generates both an In Band audible DTMF tone and an API signal causing potential confusion for devices down the line. If only the skype client had an option not the generate the tone in the first place that would be good, but then I guess they (skype) think the client would only be an end device ;-) However that is not where I'm having a problem, as I'm purely dealing with 'In band' DTMF tones. The question I had on my mind was did the Skype codec faithfully transport the DTMF tones across the network? http://fs.torr.letterboxes.org/dtmf_compare.html >From these comparisons I would have to say that there in no major filtering or distortion of the DTMF tones when transmitted across the Skype network. So I would have to say that "you can receive calls from skypeIN with inband dtmfs". If someone has a different conclusion please let me know. regards, Scott Torr On Tue, 22 Dec 2009 16:25 +0100, "Giovanni Maruzzelli" wrote: > It is probably because mod_skypiax does not analize incoming audio > looking for dtmf, because the "normal" call from a Skype client peer > sends *both* inband and out of band (signaling) dtmf. > > So, I choose to only detect out of band (signaling) dtmfs, and ignore > possible inband dtmfs (in the audio flow), so to have the most > reliable source (signaling) and spare cpu (not analizing the incoming > audio). > > Never tought you can receive calls from skypeIN with inband dtmfs... > > Open a Jira for this, I'll think about. > > Also, let me know your toughts... > > -giovanni > > > > > On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr > wrote: > > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > > FreeSWITCH Version 1.0.trunk (15787) > > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > > mod_skypiax > > > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > ? > > ? ? > > ? ? > ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > ? ? > > ? > > > > > > > > fs>console loglevel 7 > > > > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > > dtmf tones being detected and decoded by fs in the debug log. > > > > > > If however I use a pstn phone and dial my skypeIN telephone number the > > call comes into fs via skypiax but when I generate dtmf tones on the > > phone they are not detected or decoded by fs. > > > > If I take the record_session file and spectrum analyze the recorded > > tones appear to be within spec. > > > > > > Can anybody suggest why this is not working for me? > > > > > > Is the correct sample rate being used in libteletone_detect.c? > > Does the Goertzel algorithm work for other sample rates other than > > 8000hz? > > > > > > I'm not sure why I can not get this to work? > > > > > > > > regards, > > Scott Torr > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Dec 23 10:07:38 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 23 Dec 2009 19:07:38 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261591243.5459.1351510407@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <7b197bef0912220725u6ece899bo206e407198e1c350@mail.gmail.com> <1261591243.5459.1351510407@webmail.messagingengine.com> Message-ID: <7b197bef0912231007r74bc667tbe9c699679afd44@mail.gmail.com> Scott, do as tony wrote, ===== add "start_dtmf" app to your dialplan before bridge to start the inband dtmf detector. ===== -giovanni On Wed, Dec 23, 2009 at 7:00 PM, Scott Torr wrote: > Yes, > I noticed the Jira for the situation where the where the fs controlled > skype client generates both an In Band audible DTMF tone and an API > signal causing potential confusion for devices down the line. If only > the skype client had an option not the generate the tone in the first > place that would be good, but then I guess they (skype) think the client > would only be an end device ;-) > > However that is not where I'm having a problem, as I'm purely dealing > with 'In band' DTMF tones. > > The question I had on my mind was did the Skype codec faithfully > transport the DTMF tones across the network? > > http://fs.torr.letterboxes.org/dtmf_compare.html > > >From these comparisons I would have to say that there in no major > filtering or distortion of the DTMF tones when transmitted across the > Skype network. > > So I would have to say that "you can receive calls from skypeIN with > inband dtmfs". > > > If someone has a different conclusion please let me know. > > regards, > Scott Torr > > > On Tue, 22 Dec 2009 16:25 +0100, "Giovanni Maruzzelli" > wrote: >> It is probably because mod_skypiax does not analize incoming audio >> looking for dtmf, because the "normal" call from a Skype client peer >> sends *both* inband and out of band (signaling) dtmf. >> >> So, I choose to only detect out of band (signaling) dtmfs, and ignore >> possible inband dtmfs (in the audio flow), so to have the most >> reliable source (signaling) and spare cpu (not analizing the incoming >> audio). >> >> Never tought you can receive calls from skypeIN with inband dtmfs... >> >> Open a Jira for this, I'll think about. >> >> Also, let me know your toughts... >> >> -giovanni >> >> >> >> >> On Tue, Dec 22, 2009 at 3:57 PM, Scott Torr >> wrote: >> > ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> > FreeSWITCH Version 1.0.trunk (15787) >> > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> > mod_skypiax >> > >> > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> > >> > >> > ? >> > ? ? >> > ? ?> > ? ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> > ? ? >> > ? >> > >> > >> > >> > fs>console loglevel 7 >> > >> > >> > If I dial 501 from from a sip phone using "inband" dtmf I can see the >> > dtmf tones being detected and decoded by fs in the debug log. >> > >> > >> > If however I use a pstn phone and dial my skypeIN telephone number the >> > call comes into fs via skypiax but when I generate dtmf tones on the >> > phone they are not detected or decoded by fs. >> > >> > If I take the record_session file and spectrum analyze the recorded >> > tones appear to be within spec. >> > >> > >> > Can anybody suggest why this is not working for me? >> > >> > >> > Is the correct sample rate being used in libteletone_detect.c? >> > Does the Goertzel algorithm work for other sample rates other than >> > 8000hz? >> > >> > >> > I'm not sure why I can not get this to work? >> > >> > >> > >> > regards, >> > Scott Torr >> > >> > >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From scott.torr.fs at letterboxes.org Wed Dec 23 10:08:38 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 05:08:38 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> Message-ID: <1261591718.6380.1351511047@webmail.messagingengine.com> You will need to elaborate a bit more? Not sure where you want me to move the statement to? Also, In what way is a sip call handled differently to a skypiax call? Why would the sip call detect and decode properly? regards, Scott Torr On Tue, 22 Dec 2009 16:26 +0100, "Giovanni Maruzzelli" wrote: > do as anthm say :-) > > On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale > wrote: > > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf > > detector. > > > > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > > wrote: > >> > >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) > >> FreeSWITCH Version 1.0.trunk (15787) > >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > >> mod_skypiax > >> > >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > >> > >> > >> ? > >> ? ? > >> ? ? >> > >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > >> ? ? > >> ? > >> > >> > >> > >> fs>console loglevel 7 > >> > >> > >> If I dial 501 from from a sip phone using "inband" dtmf I can see the > >> dtmf tones being detected and decoded by fs in the debug log. > >> > >> > >> If however I use a pstn phone and dial my skypeIN telephone number the > >> call comes into fs via skypiax but when I generate dtmf tones on the > >> phone they are not detected or decoded by fs. > >> > >> If I take the record_session file and spectrum analyze the recorded > >> tones appear to be within spec. > >> > >> > >> Can anybody suggest why this is not working for me? > >> > >> > >> Is the correct sample rate being used in libteletone_detect.c? > >> Does the Goertzel algorithm work for other sample rates other than > >> 8000hz? > >> > >> > >> I'm not sure why I can not get this to work? > >> > >> > >> > >> regards, > >> Scott Torr > >> > >> > >> > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Dec 23 10:17:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 23 Dec 2009 19:17:21 +0100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <1261591718.6380.1351511047@webmail.messagingengine.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> <7b197bef0912220726u7f1117baie6f26b3aefe8c9c2@mail.gmail.com> <1261591718.6380.1351511047@webmail.messagingengine.com> Message-ID: <7b197bef0912231017x2f14892cr60219abda79a1971@mail.gmail.com> Ooops, Had not seen you got it in the dialplan... try to move it after the "answer" and test again. Other than this, only thing that comes in my mind is that the conversion from the pstn to sip (skype partner that gives pstn access) to skype is ruining the dtmfs beyond recognition... but you said that at spectral analisys they're fine... So, I have no idea. -giovanni On Wed, Dec 23, 2009 at 7:08 PM, Scott Torr wrote: > You will need to elaborate a bit more? > > Not sure where you want me to move the /> statement to? > > Also, > In what way is a sip call handled differently to a skypiax call? > Why would the sip call detect and decode properly? > > > ? > ? > ? > ? ? data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > ? > ? > > > > regards, > Scott Torr > > > On Tue, 22 Dec 2009 16:26 +0100, "Giovanni Maruzzelli" > wrote: >> do as anthm say :-) >> >> On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale >> wrote: >> > add "start_dtmf" app to your dialplan before bridge to start the inband dtmf >> > detector. >> > >> > >> > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr >> > wrote: >> >> >> >> ubuntu-8.04.3-server-amd64.iso (update/upgrade) >> >> FreeSWITCH Version 1.0.trunk (15787) >> >> skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb >> >> mod_skypiax >> >> >> >> (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) >> >> >> >> >> >> ? >> >> ? ? >> >> ? ?> >> >> >> ?data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> >> ? ? >> >> ? >> >> >> >> >> >> >> >> fs>console loglevel 7 >> >> >> >> >> >> If I dial 501 from from a sip phone using "inband" dtmf I can see the >> >> dtmf tones being detected and decoded by fs in the debug log. >> >> >> >> >> >> If however I use a pstn phone and dial my skypeIN telephone number the >> >> call comes into fs via skypiax but when I generate dtmf tones on the >> >> phone they are not detected or decoded by fs. >> >> >> >> If I take the record_session file and spectrum analyze the recorded >> >> tones appear to be within spec. >> >> >> >> >> >> Can anybody suggest why this is not working for me? >> >> >> >> >> >> Is the correct sample rate being used in libteletone_detect.c? >> >> Does the Goertzel algorithm work for other sample rates other than >> >> 8000hz? >> >> >> >> >> >> I'm not sure why I can not get this to work? >> >> >> >> >> >> >> >> regards, >> >> Scott Torr >> >> >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > iax:guest at conference.freeswitch.org/888 >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Wed Dec 23 10:53:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 10:53:25 -0800 Subject: [Freeswitch-users] Faxing Advice In-Reply-To: References: <87f2f3b90912010938p302d946bpeefd4631d9048aca@mail.gmail.com> <87f2f3b90912011422m4a94babah6143a048d118cb90@mail.gmail.com> Message-ID: <87f2f3b90912231053l645e841cy8a733b10c4ff1de5@mail.gmail.com> On Tue, Dec 22, 2009 at 7:58 PM, Joseph L. Casale wrote: > >> Am I correct in presuming that Freeswitch will answer a fax from a local > zap based user > >> just like it does from an FXO port connected to a POTS line? What I hope > to do here is > >> catch any call made from that extension (the zap based fax machine/user) > and push its > >> call into the fax module. > > > > Yes, when a device (phone/fax/modem/whatever) is plugged into the FXS it > gets dialtone > > and dials. Whatever it dials is put into ${destination_number} just like > any SIP phone that > > dials. This extension looks ok. Try it out and let us know how it goes. > > -MC > > Michael, > It worked well, there was however a humorous moment: I was testing with my > own shell script > that simply emailed me directly to my postfix gateway, my exchange server > and mua understood the > uuencoded attachment so once it started working I modified the script to > send to our fax service. > > Well they didn't understand uuencode so the attachment, a single page tiff, > got faxed as 23 pages > of binary :) I used mutt with a redirection to a specific muttrc which > understands mime encoding > which should work everywhere... > > Thanks for the help, you've made an office full of people happy... > Thanks for letting us know that everything worked! I'm glad we didn't have to honor Tony's triple-your-money-back guarantee. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/cb24c678/attachment-0002.html From msc at freeswitch.org Wed Dec 23 11:10:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 11:10:36 -0800 Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <26903707.post@talk.nabble.com> References: <26892767.post@talk.nabble.com> <26903707.post@talk.nabble.com> Message-ID: <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 wrote: > > More information: I can dial the default extensions like 9999 just fine. > It's > only when I call any of the IP phones (1001,1002,1003) that the call is > immediately forwarded to the callee's voice-mail when the phone goes off > the > hook. > > To only keep the SIP messages in the fs_cli screen, typing "sofia loglevel > all 0" followed by "sofia profile internal siptrace on" doesn't do the > trick, so am unable to post the whole log yet. > If you're having this much trouble with the CLI then you might be better off just using a combination of tcpdump and rotating log files. Use this command from the shell to rotate logs: fs_cli -x "fsctl send_sighup" Use the info on this page to collect a pcap: http://wiki.freeswitch.org/wiki/Packet_Capture If you have Wireshark you can open the pcap and do some fun analysis. You can also "follow the tcp stream" and watch the messages going back and forth. Hopefully you'll see what's happening (or not happening) and then we can take it from there. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/15b1124d/attachment-0002.html From larclap at yahoo.com Wed Dec 23 12:48:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 23 Dec 2009 12:48:59 -0800 Subject: [Freeswitch-users] Freeswitch not seeing Register requests In-Reply-To: <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> References: <005101ca8337$9335ffb0$b9a1ff10$@com> <87f2f3b90912221114h6b826c92w2f1125c24649c0d4@mail.gmail.com> <006c01ca833f$6a43d890$3ecb89b0$@com> <87f2f3b90912221420he1e1193g458a3fb263efdc34@mail.gmail.com> <2E0FA1A3-8740-4C43-9229-A994B026A297@jerris.com> Message-ID: <00fb01ca8411$5aa873a0$0ff95ae0$@com> Mike, You were right. I turned iptables off and the phone registered. Thanks so much, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, December 22, 2009 8:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch not seeing Register requests If your seeing the trafic in ngrep bit not in sip trace in Sofia when enabled, your firewall is blocking the traffic Mike On Dec 22, 2009, at 5:20 PM, Michael Collins wrote: On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb wrote: Yes, the internal profile exists. Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia at 192.168.10.25:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 192.168.10.25:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 192.168.10.25 alias internal ALIASED ============================================================================ ===================== 3 profiles 1 alias I would do a sanity check at this point: put this box and one phone on a completely separate network with nothing else and see what happens. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/6b1c074d/attachment-0002.html From msc at freeswitch.org Wed Dec 23 14:00:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Dec 2009 14:00:01 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call - Holiday Schedule Message-ID: <87f2f3b90912231400q65fefa87g778d1bf4857db142@mail.gmail.com> Hello all! Because the holidays fall on consecutive Fridays this year we decided to have a single conference call on Wednesday Dec 30th at the usual time of 11AM CST. The agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30 Thanks for supporting the weekly calls. Don't forget that we will soon be having giveaways and fun stuff on the calls so be sure to plan on joining us! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/5c47f7e1/attachment-0002.html From larclap at yahoo.com Wed Dec 23 17:49:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 23 Dec 2009 17:49:09 -0800 Subject: [Freeswitch-users] Local call uses public context? Message-ID: <012101ca843b$4921cfd0$db656f70$@com> I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/57f14910/attachment-0002.html From brian at freeswitch.org Wed Dec 23 18:03:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Dec 2009 20:03:28 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <012101ca843b$4921cfd0$db656f70$@com> References: <012101ca843b$4921cfd0$db656f70$@com> Message-ID: <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: > I am trying to setup a second FS box from scratch using v16048. > > What can cause a local call (81002, or 9996) to use context public? It?s a standard vanilla install. > > http://pastebin.freeswitch.org/11629 > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091223/908b5dd0/attachment-0002.html From kristoff.bonne at skypro.be Thu Dec 24 00:01:04 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Thu, 24 Dec 2009 09:01:04 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> Message-ID: <4B331FC0.1050709@skypro.be> Hi Michael, Yesterday I did a quick scan of the complete source-code of both freeswitch and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 (the two possible interface-modules to communicate with HID-devices in linux). So I fear that, at this time, it's not possible to use this device with freeswitch. :-( Cheerio! Kr. Bonne. Michael Jerris schreef: > Of course there is a way. Depending on the interface your looking at > either a freeswitch endpoiny module or an openzap module. > Mike > > On Dec 23, 2009, at 4:54 AM, Kristoff Bonne > wrote: > >> Hi Rupa, >> >> >> None. That's exactly the point. >> Everything has to be done over the usb "HID" interface. >> >> >> I've been reading about HID yesterday. HID is a usb interface that >> can be used for a large number of things, ranging from keyboard and >> game-controllers up to "water-cooling and PC-chassis" and >> point-of-sale or coin changer devices. >> >> >> It also has a telephony-interface: >> see page 69 to 72 of this document: >> http://www.usb.org/developers/devclass_docs/HID1_11.pdf >> >> This include call-control, on-hook/off-hook detection, DTMF-related >> things, etc. >> >> >> Now, the question is this: >> Is there a way to "plug" this all into freeswitch? >> >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Rupa Schomaker schreef: >>> >>> Interesting. It would have to do more than just dialtone/dtmf though. >>> Need call control, caller id, etc. What do they ship with it as far >>> as drivers go? >>> >>> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >>> wrote: >>> >>>> Hi all, >>>> >>>> >>>> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >>>> device for just 15 euro. This is a device which has on one side a >>>> USB-connector and on the other side 2 RJ-11 connectors (one FXO and >>>> one >>>> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >>>> (see here: http://www.tjnet.com/chips/tiger560C.htm) >>>> >>>> >>>> What is interesting on this device is that is uses standard USB >>>> device-classes that are by default supported by most >>>> operating-systems: >>>> usb-sound and usb-hid. >>>> >>>> >>>> When I connect it to my server (mac mini 3G running debian), the >>>> system >>>> automatically recognises these two classes >>>> >>>> [168391.922479] usbcore: registered new interface driver hiddev >>>> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID >>>> 06e6:c31c] on >>>> usb-0001:10:1b.1-1 >>>> [168391.939548] usbcore: registered new interface driver usbhid >>>> [168391.943984] usbhid: v2.6:USB HID core driver >>>> [168392.154596] usbcore: registered new interface driver snd-usb-audio >>>> >>>> >>>> And -behold- when I connect a handset in one of the port, I even get a >>>> dialtone and I can sent out DTMF-dialtone which are somehow partly >>>> (But I have no idea what program actually generates this dialtone !!!) >>>> >>>> >>>> >>>> Now, the question: >>>> Any idea if / how this can incorperated into freeswitch? Is there a >>>> way >>>> to use this device to connect a phone to freeswitch without having >>>> to go >>>> throu a SIP-client first. >>>> >>>> >>>> >>>> Cheerio! Kr. Bonne. >>>> >>>> -- >>>> jabber/gtalk: kristoff at krbonne.net >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> >>> >>> >>> >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6def3e76/attachment-0002.bin From gmaruzz at celliax.org Thu Dec 24 00:09:44 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 24 Dec 2009 09:09:44 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B331FC0.1050709@skypro.be> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> <4B331FC0.1050709@skypro.be> Message-ID: <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> Hi Kristoff, if you can send me infos on where to buy it, I probably can adapt my upcoming mod_gsmopen endpoint to manage your device... -giovanni On Thu, Dec 24, 2009 at 9:01 AM, Kristoff Bonne wrote: > Hi Michael, > > > > Yesterday I did a quick scan of the complete source-code of both freeswitch > and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 > (the two possible interface-modules to communicate with HID-devices in > linux). > > So I fear that, at this time, it's not possible to use this device with > freeswitch. :-( > > > > Cheerio! Kr. Bonne. > > > Michael Jerris schreef: > > Of course there is a way.? Depending on the interface your looking at either > a freeswitch endpoiny module or an openzap module. > Mike > > On Dec 23, 2009, at 4:54 AM, Kristoff Bonne > wrote: > > Hi Rupa, > > > None. That's exactly the point. > Everything has to be done over the usb "HID" interface. > > > I've been reading about HID yesterday. HID is a usb interface that can be > used for a large number of things, ranging from keyboard and > game-controllers up to "water-cooling and PC-chassis" and point-of-sale or > coin changer devices. > > > It also has a telephony-interface: > see page 69 to 72 of this document: > http://www.usb.org/developers/devclass_docs/HID1_11.pdf > > This include call-control, on-hook/off-hook detection, DTMF-related things, > etc. > > > Now, the question is this: > Is there a way to "plug" this all into freeswitch? > > > > > Cheerio! Kr. Bonne. > > > Rupa Schomaker schreef: > > Interesting.? It would have to do more than just dialtone/dtmf though. > ?Need call control, caller id, etc.? What do they ship with it as far > as drivers go? > > On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne > wrote: > > Hi all, > > > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) > > > What is interesting on this device is that is uses standard USB > device-classes that are by default supported by most operating-systems: > usb-sound and usb-hid. > > > When I connect it to my server (mac mini 3G running debian), the system > automatically recognises these two classes > > [168391.922479] usbcore: registered new interface driver hiddev > [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on > usb-0001:10:1b.1-1 > [168391.939548] usbcore: registered new interface driver usbhid > [168391.943984] usbhid: v2.6:USB HID core driver > [168392.154596] usbcore: registered new interface driver snd-usb-audio > > > And -behold- when I connect a handset in one of the port, I even get a > dialtone and I can sent out DTMF-dialtone which are somehow partly > (But I have no idea what program actually generates this dialtone !!!) > > > > Now, the question: > Any idea if / how this can incorperated into freeswitch? Is there a way > to use this device to connect a phone to freeswitch without having to go > throu a SIP-client first. > > > > Cheerio! Kr. Bonne. > > -- > jabber/gtalk: kristoff at krbonne.net > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > jabber/gtalk: kristoff at krbonne.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From kristoff.bonne at skypro.be Thu Dec 24 00:22:31 2009 From: kristoff.bonne at skypro.be (Kristoff Bonne) Date: Thu, 24 Dec 2009 09:22:31 +0100 Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> References: <4B3142F7.1080600@skypro.be> <4B31E8F2.6040907@skypro.be> <212F33E8-855E-4695-A5C7-3F269B955D12@jerris.com> <4B331FC0.1050709@skypro.be> <7b197bef0912240009j540b231cg99bb4570df76086d@mail.gmail.com> Message-ID: <4B3324C7.7090804@skypro.be> Hi Giovanni, I actually got if from my local "carrefour" supermarket (you know, the french supermarket-chain). The product they sell is this: http://www.profoon.nl/TradePoint/Item_View?itemNo=IP-150 It's called a "skype-gateway" (as you can use it with the skype softphone too). Cheerio! Kr. Bonne. Giovanni Maruzzelli schreef: > Hi Kristoff, > > if you can send me infos on where to buy it, I probably can adapt my > upcoming mod_gsmopen endpoint to manage your device... > > -giovanni > > On Thu, Dec 24, 2009 at 9:01 AM, Kristoff Bonne > wrote: > >> Hi Michael, >> >> >> >> Yesterday I did a quick scan of the complete source-code of both freeswitch >> and openzap and there is nowhere any mention of libhid or /dev/usb/hiddev0 >> (the two possible interface-modules to communicate with HID-devices in >> linux). >> >> So I fear that, at this time, it's not possible to use this device with >> freeswitch. :-( >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Michael Jerris schreef: >> >> Of course there is a way. Depending on the interface your looking at either >> a freeswitch endpoiny module or an openzap module. >> Mike >> >> On Dec 23, 2009, at 4:54 AM, Kristoff Bonne >> wrote: >> >> Hi Rupa, >> >> >> None. That's exactly the point. >> Everything has to be done over the usb "HID" interface. >> >> >> I've been reading about HID yesterday. HID is a usb interface that can be >> used for a large number of things, ranging from keyboard and >> game-controllers up to "water-cooling and PC-chassis" and point-of-sale or >> coin changer devices. >> >> >> It also has a telephony-interface: >> see page 69 to 72 of this document: >> http://www.usb.org/developers/devclass_docs/HID1_11.pdf >> >> This include call-control, on-hook/off-hook detection, DTMF-related things, >> etc. >> >> >> Now, the question is this: >> Is there a way to "plug" this all into freeswitch? >> >> >> >> >> Cheerio! Kr. Bonne. >> >> >> Rupa Schomaker schreef: >> >> Interesting. It would have to do more than just dialtone/dtmf though. >> Need call control, caller id, etc. What do they ship with it as far >> as drivers go? >> >> On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne >> wrote: >> >> Hi all, >> >> >> This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >> device for just 15 euro. This is a device which has on one side a >> USB-connector and on the other side 2 RJ-11 connectors (one FXO and one >> FSX). Internally, the device seams to contain a tigerjet 560C chipset. >> (see here: http://www.tjnet.com/chips/tiger560C.htm) >> >> >> What is interesting on this device is that is uses standard USB >> device-classes that are by default supported by most operating-systems: >> usb-sound and usb-hid. >> >> >> When I connect it to my server (mac mini 3G running debian), the system >> automatically recognises these two classes >> >> [168391.922479] usbcore: registered new interface driver hiddev >> [168391.935068] hiddev0hidraw0: USB HID v1.00 Device [HID 06e6:c31c] on >> usb-0001:10:1b.1-1 >> [168391.939548] usbcore: registered new interface driver usbhid >> [168391.943984] usbhid: v2.6:USB HID core driver >> [168392.154596] usbcore: registered new interface driver snd-usb-audio >> >> >> And -behold- when I connect a handset in one of the port, I even get a >> dialtone and I can sent out DTMF-dialtone which are somehow partly >> (But I have no idea what program actually generates this dialtone !!!) >> >> >> >> Now, the question: >> Any idea if / how this can incorperated into freeswitch? Is there a way >> to use this device to connect a phone to freeswitch without having to go >> throu a SIP-client first. >> >> >> >> Cheerio! Kr. Bonne. >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> ________________________________ >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> jabber/gtalk: kristoff at krbonne.net >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > -- jabber/gtalk: kristoff at krbonne.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: kristoff_bonne.vcf Type: text/x-vcard Size: 190 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment-0002.vcf -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/9ceccdef/attachment-0002.bin From mcampbellsmith at gmail.com Thu Dec 24 03:16:31 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 24 Dec 2009 22:16:31 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> Message-ID: <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> Sorry for the delay. Christmas stuff! I have put the log at http://pastebin.freeswitch.org/11632 The SPA3102 has remote IP address 11.11.11.11 (in the log), and internal ip address 192.168.1.3 My local ATA has external IP address 124.11.11.11 (in the log) and internal ip address 192.168.1.120 I have all these set: > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server: stun.freeswitch.org When I set 'Nat Mapping Enable' under tab Line 1 in the SPA3102, I get the following trace. This is all I see and then registration fails. freeswitch at internal> recv 545 bytes from tls/[11.11.11.11]:56886 at 11:12:22.924395: ------------------------------------------------------------------------ REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport From: 2001 ;tag=2f998bc591c1321o0 To: 2001 Call-ID: 115e0ffa-a538d31b at 192.168.1.3 CSeq: 35409 REGISTER Max-Forwards: 70 Contact: 2001 ;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces ------------------------------------------------------------------------ send 678 bytes to tls/[11.11.11.11]:56886 at 11:12:22.972012: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport=56886;received=11.11.11.11 From: 2001 ;tag=2f998bc591c1321o0 To: 2001 ;tag=07c0ymHUv7KXH Call-ID: 115e0ffa-a538d31b at 192.168.1.3 CSeq: 35409 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15490 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="myddns.dydns.org", nonce="35a9849e-f07d-11de-88a5-dbc3ffce4ce8", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ ------------------------------------------------------------------------ Thanks! On Tue, Dec 22, 2009 at 2:03 AM, Brian West wrote: > Can you get me siptraces please. > /b > On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote: > > Thanks Brian and Gad, > > I have stun set and if I do a 'sofia status profile internal', I see > the external IP address of the 3102 ATA, so I assume that stun is > working correctly on the SPA3102. > > These are the options that I have set (according to the 3102 manual). > > ? Handle VIA received: yes > ? Handle VIA rport: yes > ? Insert VIA received: yes > ? Insert VIA rport: yes > ? Substitute VIA Addr: yes > ? Send Resp To Src Port: yes > ? STUN Enable: Choose yes. > ? STUN Server:?stun.freeswitch.org > > I assume that is all is needed? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From scott.torr.fs at letterboxes.org Thu Dec 24 04:29:10 2009 From: scott.torr.fs at letterboxes.org (Scott Torr) Date: Thu, 24 Dec 2009 23:29:10 +1100 Subject: [Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call? In-Reply-To: <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> References: <1261493825.21085.1351311647@webmail.messagingengine.com> <191c3a030912220721x53126c61s62fe8b85418657f8@mail.gmail.com> Message-ID: <1261657750.2312.1351608571@webmail.messagingengine.com> Hi Anthony, Yes, The "start_dtmf" application is in the dialplan. One question I still have is will the Goertzel algorithm in libteletone_detect.c be able to detect and decode the DTMF tones once they have past through the PSTN and Skype network traversing various codecs? 1) They sound audible and clear. 2) A spectrum graph clearly shows the two frequencies. How bad does the signal need to degrade before the DTMF tones cannot be detected? Can you suggest a way to play recordings through the "start_dtmf" application. This way I can test various wave forms. ** BUG ** Why does samples=0? One thing I have noted is that when "start_ivr_async.c" calls: teletone_dtmf_detect(&pvt->dtmf_detect, frame->data, frame->samples); for a skypiax call the samples=0 for a SIP call the samples=160 I hope this may help track down the problem. Perhaps in time with better understanding of the internal workings of fs and may be able to post solutions rather than problems? regards, Scott Torr On Tue, 22 Dec 2009 09:21 -0600, "Anthony Minessale" wrote: > add "start_dtmf" app to your dialplan before bridge to start the inband > dtmf > detector. > > > On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr > wrote: > > > ubuntu-8.04.3-server-amd64.iso (update/upgrade) > > FreeSWITCH Version 1.0.trunk (15787) > > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb > > mod_skypiax > > > > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs) > > > > > > > > > > > > > data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > > > > > > > fs>console loglevel 7 > > > > > > If I dial 501 from from a sip phone using "inband" dtmf I can see the > > dtmf tones being detected and decoded by fs in the debug log. > > > > > > If however I use a pstn phone and dial my skypeIN telephone number the > > call comes into fs via skypiax but when I generate dtmf tones on the > > phone they are not detected or decoded by fs. > > > > If I take the record_session file and spectrum analyze the recorded > > tones appear to be within spec. > > > > > > Can anybody suggest why this is not working for me? > > > > > > Is the correct sample rate being used in libteletone_detect.c? > > Does the Goertzel algorithm work for other sample rates other than > > 8000hz? > > > > > > I'm not sure why I can not get this to work? > > > > > > > > regards, > > Scott Torr > > > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 From nicolas at medularis.com Thu Dec 24 05:05:18 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 10:05:18 -0300 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! Message-ID: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> I just got into the fs cli and when I ran a 'show calls' I got the following message: 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! What does this mean? does it mean the event_socket did not report 8456 events? Why could this happen? The answer to this is pretty critical to me, as I make and monitor calls through the socket. Thanks for your help! Nicolas From yehavi.bourvine at gmail.com Thu Dec 24 06:35:20 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Dec 2009 16:35:20 +0200 Subject: [Freeswitch-users] SNOM shared lines with TLS problems? Message-ID: Hello, Is there anyone who is using SNOM with TLS encryption and shared lines and it works? We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: - With non-shared line UDP and TLS both work ok. - With shared lines UDP works ok. - with shared line TLS works as long as only one phone is registered. - After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. Anyone has this setup working and can share some tips? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/c5fb4a1e/attachment-0002.html From nicolas at medularis.com Thu Dec 24 07:19:45 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 12:19:45 -0300 Subject: [Freeswitch-users] Javascript system calls Message-ID: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> Hi, I wanted to know what is the javascript equivalent of lua's os.execute(). I need to run a command from within a js script. Thanks! Nicolas From nicolas at medularis.com Thu Dec 24 07:37:52 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 24 Dec 2009 12:37:52 -0300 Subject: [Freeswitch-users] Javascript system calls In-Reply-To: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> References: <1b46b4e80912240719s1267f262v84436000228f5d48@mail.gmail.com> Message-ID: <1b46b4e80912240737n56ab173fg70fadb50935397f@mail.gmail.com> I'll reply to myself: the function system() On Thu, Dec 24, 2009 at 12:19 PM, Nicolas Brenner wrote: > Hi, I wanted to know what is the javascript equivalent of lua's > os.execute(). I need to run a command from within a js script. > > Thanks! > > Nicolas > From larclap at yahoo.com Thu Dec 24 08:16:53 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 24 Dec 2009 08:16:53 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> Message-ID: <002901ca84b4$819f9760$84dec620$@com> Brian, Please forgive my slowness, but I'm still having problems with this. When you say that I "really didn't auth the user", did you mean the endpoint/extension? If you did, I upped to svn1 16055 and placed a cidr attribute on the extension and reran the test, resulting in the same output, going to context public. Further, I'm confused about your response about ACL compared with Billy W in an email of 12/22/2009. ".you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from." http://pastebin.freeswitch.org/11633 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 23, 2009 6:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/e23c0dab/attachment-0002.html From msc at freeswitch.org Thu Dec 24 10:59:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Dec 2009 10:59:43 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <002901ca84b4$819f9760$84dec620$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> Message-ID: <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> Lars, Since this question has come up a few times I'm going to write up a nice wiki article on it explaining the differences between letting someone in via an ACL and actually doing digest authentication. In a nutshell, though, it's this: if the user does digest authentication (with the whole REGISTER, 401, REGISTER, 200 OK exchange) then whatever value is in user_context is the context for the calls made by that user. In conf/directory/default/1000.xml (and 1001.xml, etc.) they all have user_context = "default" so when those users register the calls they make are handled in the default context. OTOH, if you let a user in via an ACL they aren't really registered, you've simply opened the door for anyone coming from a particular IP address or IP address range. In that case the calls are handled in the context specified by the context parameter of the sip profile where the calls come in. By default the internal sip profile uses the public context. This is for security reasons. "Paranoid by default" is how you might describe it. You are welcome to change that value to "default" so that calls let in by the ACL are handled just like auth'd calls. Play around with it and let us know how it goes. I think you'll get it once you start modifying settings and making test calls. -MC On Thu, Dec 24, 2009 at 8:16 AM, Lars Zeb wrote: > Brian, > > > > Please forgive my slowness, but I?m still having problems with this. When > you say that I ?really didn?t auth the user?, did you mean the > endpoint/extension? > > > > If you did, I upped to svn1 16055 and placed a cidr attribute on the > extension and reran the test, resulting in the same output, going to context > public. > > > > Further, I?m confused about your response about ACL compared with Billy W > in an email of 12/22/2009. > > > > ??you could simply put these entries in your internal sofia profile. > > > > name="apply-register-acl" value="192.168.0.0/24"/> > > > > In that case, you do not need to include anything in the directory. The > cidr entries in the directory are for providing additional control for each > user id and what IPs they are allowed to make calls from.? > > > > http://pastebin.freeswitch.org/11633 > > Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 > i386 GNU/Linux > > > > Thanks Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, December 23, 2009 6:03 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Local call uses public context? > > > > 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved > by acl "192.168.10.0/24[]". Access Granted. > > > > Because the context is set on the profile as public... and you really > didn't auth the user so user_context was never set. > > > > /b > > > > On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: > > > > I am trying to setup a second FS box from scratch using v16048. > > > > What can cause a local call (81002, or 9996) to use context public? It?s a > standard vanilla install. > > > > http://pastebin.freeswitch.org/11629 > > > > Thanks, Lars > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/16ccfd9f/attachment-0002.html From larclap at yahoo.com Thu Dec 24 12:31:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 24 Dec 2009 12:31:51 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> Message-ID: <000901ca84d8$1fee8c50$5fcba4f0$@com> Thanks for the reply, Michael. I tried the digest authentication using the cidr and copying the conf/sip_profiles/internal.xml from the distribution, where As a result, one endpoint could not register and another was unauthorized. http://pastebin.freeswitch.org/11634 Then I went changed the context in internal.xml from public to default and And the phones registered OK. So my confusion persists. Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 24, 2009 11:00 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? Lars, Since this question has come up a few times I'm going to write up a nice wiki article on it explaining the differences between letting someone in via an ACL and actually doing digest authentication. In a nutshell, though, it's this: if the user does digest authentication (with the whole REGISTER, 401, REGISTER, 200 OK exchange) then whatever value is in user_context is the context for the calls made by that user. In conf/directory/default/1000.xml (and 1001.xml, etc.) they all have user_context = "default" so when those users register the calls they make are handled in the default context. OTOH, if you let a user in via an ACL they aren't really registered, you've simply opened the door for anyone coming from a particular IP address or IP address range. In that case the calls are handled in the context specified by the context parameter of the sip profile where the calls come in. By default the internal sip profile uses the public context. This is for security reasons. "Paranoid by default" is how you might describe it. You are welcome to change that value to "default" so that calls let in by the ACL are handled just like auth'd calls. Play around with it and let us know how it goes. I think you'll get it once you start modifying settings and making test calls. -MC On Thu, Dec 24, 2009 at 8:16 AM, Lars Zeb wrote: Brian, Please forgive my slowness, but I'm still having problems with this. When you say that I "really didn't auth the user", did you mean the endpoint/extension? If you did, I upped to svn1 16055 and placed a cidr attribute on the extension and reran the test, resulting in the same output, going to context public. Further, I'm confused about your response about ACL compared with Billy W in an email of 12/22/2009. ".you could simply put these entries in your internal sofia profile. In that case, you do not need to include anything in the directory. The cidr entries in the directory are for providing additional control for each user id and what IPs they are allowed to make calls from." http://pastebin.freeswitch.org/11633 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, December 23, 2009 6:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Local call uses public context? 2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by acl "192.168.10.0/24[]". Access Granted. Because the context is set on the profile as public... and you really didn't auth the user so user_context was never set. /b On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote: I am trying to setup a second FS box from scratch using v16048. What can cause a local call (81002, or 9996) to use context public? It's a standard vanilla install. http://pastebin.freeswitch.org/11629 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/bd0f5417/attachment-0002.html From ken at ukgb.net Thu Dec 24 12:42:32 2009 From: ken at ukgb.net (Ken Gillett) Date: Thu, 24 Dec 2009 20:42:32 +0000 Subject: [Freeswitch-users] MacOSX Message-ID: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> Yes, I want to set up FreeSwitch on OSX and at least see how it runs, assuming I can get that far. I've downloaded the latest tarball and run configure which seemed to complete ok. But what next? Do I actually need to run make all install sounds-install moh-install as in some lists of instructions it appears that I just need to run make make install Needless to say I'm not an expert at compiling although I have done a fair bit over the years, just not enough for it to be second nature. So the above apparent ambiguity puzzles me. Also, how can I compile on one machine and then actually run it on a different machine? Is there a relatively simple way to achieve this or must I manually copy all the files to the other machine. What files would that be? Are they all conveniently located in a single folder? Hopeful of some helpful advice, but let's face it, anyone doing this sort of thing on Christmas Eve really ought to get out more:-) Ken G i l l e t t _/_/_/_/_/_/_/_/ From jaugenstine at gmail.com Thu Dec 24 20:40:04 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 24 Dec 2009 20:40:04 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> Message-ID: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Ken, The process is the same for all UNIX like platforms. You run - bootstrap.sh - configure - make - make install Jonathan On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: > Yes, I want to set up FreeSwitch on OSX and at least see how it runs, > assuming I can get that far. > > I've downloaded the latest tarball and run configure which seemed to > complete ok. But what next? Do I actually need to run > > make all install sounds-install moh-install > > as in some lists of instructions it appears that I just need to run > > make > make install > > Needless to say I'm not an expert at compiling although I have done a fair > bit over the years, just not enough for it to be second nature. So the above > apparent ambiguity puzzles me. > > Also, how can I compile on one machine and then actually run it on a > different machine? Is there a relatively simple way to achieve this or must > I manually copy all the files to the other machine. What files would that > be? Are they all conveniently located in a single folder? > > Hopeful of some helpful advice, but let's face it, anyone doing this sort > of thing on Christmas Eve really ought to get out more:-) > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091224/6df46cb2/attachment-0002.html From jason at jasonjgw.net Thu Dec 24 21:23:51 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 25 Dec 2009 16:23:51 +1100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <20091225052351.GA8608@jdc.jasonjgw.net> jonathan augenstine wrote: > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install Unless there are package files (as in .deb and .rpm) for your operating system that can be generated from the sources, in which case you should run the appropriate package building tools. I always prefer to use the package management system, where possible, instead of just compiling and installing software into /usr/local. From ken at ukgb.net Fri Dec 25 01:07:20 2009 From: ken at ukgb.net (Ken Gillett) Date: Fri, 25 Dec 2009 09:07:20 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: What if you've edited modules.conf and why do the install instructions say to run make all install sounds-install moh-install. Also, what is bootstrap.sh? I see it mentioned in some places in the instructions, but not others. Why might I need to run it? On 25 Dec 2009, at 04:40, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan Ken G i l l e t t _/_/_/_/_/_/_/_/ From ken at ukgb.net Fri Dec 25 01:14:31 2009 From: ken at ukgb.net (Ken Gillett) Date: Fri, 25 Dec 2009 09:14:31 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <20091225052351.GA8608@jdc.jasonjgw.net> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <20091225052351.GA8608@jdc.jasonjgw.net> Message-ID: I'd liked to create an install package, but that's a whole new can of worms. Are the FreeSwitch files all installed in a single directory that could be copied to a different machine? On 25 Dec 2009, at 05:23, Jason White wrote: > I always prefer to use the package management system, where possible, instead > of just compiling and installing software into /usr/local. > Ken G i l l e t t _/_/_/_/_/_/_/_/ From qinglan_zeng at hotmail.com Fri Dec 25 01:20:44 2009 From: qinglan_zeng at hotmail.com (=?gb2312?B?tPPE4MjL?=) Date: Fri, 25 Dec 2009 09:20:44 +0000 Subject: [Freeswitch-users] GSM modem for Skype In-Reply-To: References: Message-ID: Hello All, I had a GSM VoIP Gateway which can be connected to service providers who use SIP protocal. I remember somebody in Freeswitch community mentioned there is a software which can connect GSM network with Skype network. I'm not sure if this software can work with my GSM VOIP gateway or not. If somebody can send me such software that would be much appriciated and even license fee required would be accepted. I hope I can have a testing on this or I can send this gateway to those guys who have this kind of software for testing. Thanks and Merry Christmas. Daniel Zeng _________________________________________________________________ ?????????????????msn????? http://ditu.live.com/?form=TL&swm=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/6cf83f27/attachment-0002.html From jason at jasonjgw.net Fri Dec 25 01:43:55 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 25 Dec 2009 20:43:55 +1100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <20091225052351.GA8608@jdc.jasonjgw.net> Message-ID: <20091225094355.GA11565@jdc.jasonjgw.net> Ken Gillett wrote: > I'd liked to create an install package, but that's a whole new can of worms. > > Are the FreeSwitch files all installed in a single directory that could be > copied to a different machine? Yes, but that isn't a substitute for package management. For example, you typically want to preserve configuration files across package updates while having the opportunity to merge changes from newer configuration files. Package management solves this problem; it also solves dependency issues. From lei.tlfly at gmail.com Fri Dec 25 04:36:33 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 25 Dec 2009 20:36:33 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected Message-ID: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> Hi all and merry holiday, I have encounter fs core dump many times when I exit fs_cli, I'm using the fs 1.0.5pre9. I can reproduce this fault by follow steps 1.launch fs with console 2.press ctrl+z to ext from fs console 3.run fs_cli (from local) 4.press ctrl+z to exit fs_cli (or type /bye) 5.fs core dump. This fault is not reproduced every times, but quite frequent. does some have encountered the same problem or have any idea about it? Any suggestion is appreciated! ====here is the back traces in gdb #0 0x02a6593c in ?? () (gdb) where #0 0x02a6593c in ?? () #1 #2 0x0028a410 in __kernel_vsyscall () #3 0x0075870b in write () from /lib/libpthread.so.0 #4 0x0035451a in apr_socket_send (sock=0x85a12f8, buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0x10aaa78) at network_io/unix/sendrecv.c:41 #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) at /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at threadproc/unix/thread.c:138 ====another back trace (gdb) where #0 0x075e893c in ?? () #1 #2 0x002ae410 in __kernel_vsyscall () #3 0x0075870b in write () from /lib/libpthread.so.0 #4 0x001d951a in apr_socket_send (sock=0x92f8650, buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0xe9ea78) at network_io/unix/sendrecv.c:41 #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) at /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at threadproc/unix/thread.c:138 #8 0x007515ab in start_thread () from /lib/libpthread.so.0 #9 0x006a7cfe in clone () from /lib/libc.so.6 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/e74380a9/attachment-0002.html From lei.tlfly at gmail.com Fri Dec 25 04:37:52 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Fri, 25 Dec 2009 20:37:52 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected In-Reply-To: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> References: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> Message-ID: <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> BTW my is environment [root at localhost bin]# uname -a Linux localhost.localdomain 2.6.18-164.el5PAE #1 SMP Thu Sep 3 04:10:44 EDT 2009 i686 i686 i386 GNU/Linux [root at localhost bin]# gcc -v ???? specs? ???i386-redhat-linux ????../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk --disable-dssi --enable-plugin --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic --host=i386-redhat-linux ?????posix gcc ?? 4.1.2 20080704 (Red Hat 4.1.2-46) 2009/12/25 Lei Tang > Hi all and merry holiday, I have encounter fs core dump many times when I > exit fs_cli, I'm using the fs 1.0.5pre9. > I can reproduce this fault by follow steps > 1.launch fs with console > 2.press ctrl+z to ext from fs console > 3.run fs_cli (from local) > 4.press ctrl+z to exit fs_cli (or type /bye) > 5.fs core dump. > > This fault is not reproduced every times, but quite frequent. does some > have encountered the same problem or have any idea about it? > Any suggestion is appreciated! > > > ====here is the back traces in gdb > #0 0x02a6593c in ?? () > (gdb) where > #0 0x02a6593c in ?? () > #1 > #2 0x0028a410 in __kernel_vsyscall () > #3 0x0075870b in write () from /lib/libpthread.so.0 > #4 0x0035451a in apr_socket_send (sock=0x85a12f8, > buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0x10aaa78) > at network_io/unix/sendrecv.c:41 > #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, > buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 > #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) > at > /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 > #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at > threadproc/unix/thread.c:138 > > ====another back trace > (gdb) where > #0 0x075e893c in ?? () > #1 > #2 0x002ae410 in __kernel_vsyscall () > #3 0x0075870b in write () from /lib/libpthread.so.0 > #4 0x001d951a in apr_socket_send (sock=0x92f8650, > buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0xe9ea78) > at network_io/unix/sendrecv.c:41 > #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, > buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! > http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 > #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) > at > /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 > #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at > threadproc/unix/thread.c:138 > #8 0x007515ab in start_thread () from /lib/libpthread.so.0 > #9 0x006a7cfe in clone () from /lib/libc.so.6 > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/2153e486/attachment-0002.html From jbr at consiglia.dk Fri Dec 25 09:25:23 2009 From: jbr at consiglia.dk (Jon Bruel) Date: Fri, 25 Dec 2009 18:25:23 +0100 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: I have added an example on the wiki illustrating how to propagate presence and registrations over a set of networked FSs. Interested? Find it on: http://wiki.freeswitch.org/wiki/Mod_event_multicast. /Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/61f23e5b/attachment-0002.html From xengelpublicx at gmail.com Fri Dec 25 10:53:47 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Fri, 25 Dec 2009 21:53:47 +0300 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: <4B350A3B.3060305@gmail.com> On 25.12.2009 20:25, Jon Bruel wrote: > I have added an example on the wiki illustrating how to propagate > presence and registrations over a set of networked FSs. Interested? Yes. From rob4manhere at gmail.com Fri Dec 25 11:41:31 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Fri, 25 Dec 2009 13:41:31 -0600 Subject: [Freeswitch-users] Presence across several networked FSs In-Reply-To: References: Message-ID: <462386EC-8D78-411F-BD30-E42317915041@gmail.com> Very nice- thanks Jon! On Dec 25, 2009, at 11:25 AM, Jon Bruel wrote: > I have added an example on the wiki illustrating how to propagate > presence and registrations over a set of networked FSs. Interested? > Find it on:http://wiki.freeswitch.org/wiki/Mod_event_multicast. /Jon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091225/6130eb92/attachment-0002.html From neilp at cs.stanford.edu Sat Dec 26 01:29:58 2009 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 26 Dec 2009 01:29:58 -0800 Subject: [Freeswitch-users] freeswitch init Message-ID: Hi All, On the Freeswitch init page (Debian) the setup you to create a user "freeswitch" to run the FS process. When I did this, freeswitch started up but wasn't able to find/open channels to my sangoma/wanpipe hardware. Is this because: 1. The sangoma hardware was installed and running through root, which is a different user than the one running FS? 2. Wanrouter is not getting started before freeswitch on system boot? Is there a disadvantage to setting up the init script for FS to be run by root? Will that solve this problem? Thanks in advance from a Linux/FS newbie, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/1cc56b24/attachment-0002.html From jason at jasonjgw.net Sat Dec 26 02:05:26 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 26 Dec 2009 21:05:26 +1100 Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <20091226100526.GA13489@jdc.jasonjgw.net> Neil Patel wrote: > On the Freeswitch init page (Debian) the setup you to create a user > "freeswitch" to run the FS process. When I did this, freeswitch started up > but wasn't able to find/open channels to my sangoma/wanpipe hardware. It's likely to be the result of wrong permissions on the device file, or maybe you need to add the freeswitch user to a group that has permission to access the device file. I don't have access to Sangoma hardware, so I don't know what the device files are, but this should be documented somewhere, or perhaps it is noted in your kernel logs. From info at daccii.it Sat Dec 26 02:33:58 2009 From: info at daccii.it (Daniele Salvatore Albano) Date: Sat, 26 Dec 2009 11:33:58 +0100 Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <4B35E696.3090408@daccii.it> Hi, you should take a look to /etc/udev/rules.d searching for a zaptel.[something] (extension should be .conf, but i'm not sure). Open it and change user and group in to freeswitch. To change the startup/shutdown sequence, instead, you could use update-rc.d. First look at startup and shutdown order for wanrouter and after execute these commands: sudo update-rc.d -f freeswitch remove sudo update-rc.d freeswitch defaults SS KK On SS put a value greater than wanrouter startup order value and on KK put a value freater than warouter shutdown order value. As "shutdown/startup order value" i refer to SNNxxxxxx or KNNxxxxxx where NN is a numeric value and xxxxxx is the name of the service. You should look for them into /etc/rc3.d and /etc/rc0.d (the first for startup and the second for shutdown) Neil Patel ha scritto: > Hi All, > > On the Freeswitch init page (Debian) the setup you to create a user > "freeswitch" to run the FS process. When I did this, freeswitch > started up but wasn't able to find/open channels to my sangoma/wanpipe > hardware. Is this because: > > 1. The sangoma hardware was installed and running through root, which > is a different user than the one running FS? > 2. Wanrouter is not getting started before freeswitch on system boot? > > Is there a disadvantage to setting up the init script for FS to be run > by root? Will that solve this problem? > > Thanks in advance from a Linux/FS newbie, > Neil -------------- next part -------------- A non-text attachment was scrubbed... Name: info.vcf Type: text/x-vcard Size: 307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/95d08164/attachment-0002.vcf From lei.tlfly at gmail.com Sat Dec 26 03:48:57 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Sat, 26 Dec 2009 19:48:57 +0800 Subject: [Freeswitch-users] fs core dump after fs_cli disconnected In-Reply-To: <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> References: <50c41b4e0912250436w2b5267b8rbbdce6e5b2cc212b@mail.gmail.com> <50c41b4e0912250437q78f2bc12t75f079dc1c730ee3@mail.gmail.com> Message-ID: <50c41b4e0912260348k23ed56bdk3f34f9e3113179b3@mail.gmail.com> Hi all, I have found the cause of this problem. It due to some code in a library I loaded into Fs, it set SIGPIPE handler, the handler seemed to be invalid when SIGPIPE is fired, so FS is broken. address, so, 2009/12/25 Lei Tang > BTW my is environment > > [root at localhost bin]# uname -a > Linux localhost.localdomain 2.6.18-164.el5PAE #1 SMP Thu Sep 3 04:10:44 EDT > 2009 i686 i686 i386 GNU/Linux > [root at localhost bin]# gcc -v > ???? specs? > ???i386-redhat-linux > ????../configure --prefix=/usr --mandir=/usr/share/man > --infodir=/usr/share/info --enable-shared --enable-threads=posix > --enable-checking=release --with-system-zlib --enable-__cxa_atexit > --disable-libunwind-exceptions --enable-libgcj-multifile > --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk > --disable-dssi --enable-plugin > --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --with-cpu=generic > --host=i386-redhat-linux > ?????posix > gcc ?? 4.1.2 20080704 (Red Hat 4.1.2-46) > > > 2009/12/25 Lei Tang > > Hi all and merry holiday, I have encounter fs core dump many times when I >> exit fs_cli, I'm using the fs 1.0.5pre9. >> I can reproduce this fault by follow steps >> 1.launch fs with console >> 2.press ctrl+z to ext from fs console >> 3.run fs_cli (from local) >> 4.press ctrl+z to exit fs_cli (or type /bye) >> 5.fs core dump. >> >> This fault is not reproduced every times, but quite frequent. does some >> have encountered the same problem or have any idea about it? >> Any suggestion is appreciated! >> >> >> ====here is the back traces in gdb >> #0 0x02a6593c in ?? () >> (gdb) where >> #0 0x02a6593c in ?? () >> #1 >> #2 0x0028a410 in __kernel_vsyscall () >> #3 0x0075870b in write () from /lib/libpthread.so.0 >> #4 0x0035451a in apr_socket_send (sock=0x85a12f8, >> buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0x10aaa78) >> at network_io/unix/sendrecv.c:41 >> #5 0x002b8d11 in switch_socket_send (sock=0x85a12f8, >> buf=0x10ab2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0x10ab368) at src/switch_apr.c:697 >> #6 0x0011a3b2 in listener_run (thread=0xb6d9bbf8, obj=0x85a1490) >> at >> /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 >> #7 0x003564f6 in dummy_worker (opaque=0xb6d9bbf8) at >> threadproc/unix/thread.c:138 >> >> ====another back trace >> (gdb) where >> #0 0x075e893c in ?? () >> #1 >> #2 0x002ae410 in __kernel_vsyscall () >> #3 0x0075870b in write () from /lib/libpthread.so.0 >> #4 0x001d951a in apr_socket_send (sock=0x92f8650, >> buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0xe9ea78) >> at network_io/unix/sendrecv.c:41 >> #5 0x0013dd11 in switch_socket_send (sock=0x92f8650, >> buf=0xe9f2fb "Disconnected, goodbye.\nSee you at ClueCon! >> http://www.cluecon.com/\n", len=0xe9f368) at src/switch_apr.c:697 >> #6 0x005473b2 in listener_run (thread=0xb6dfebf8, obj=0x92f87e8) >> at >> /root/src/freeswitch-1.0.5pre9/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2188 >> #7 0x001db4f6 in dummy_worker (opaque=0xb6dfebf8) at >> threadproc/unix/thread.c:138 >> #8 0x007515ab in start_thread () from /lib/libpthread.so.0 >> #9 0x006a7cfe in clone () from /lib/libc.so.6 >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/5506a45d/attachment-0002.html From max.bridgewater at gmail.com Sat Dec 26 19:51:34 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Sat, 26 Dec 2009 22:51:34 -0500 Subject: [Freeswitch-users] No gsmopen in trunk? Message-ID: Hi, I just did a fresh checkout of Freeswitch from trunk but it seems mod_gsmopen is not there. Am I doing something weird or it has been removed from trunk. Please advise! Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091226/ccece4dd/attachment-0002.html From yehavi.bourvine at gmail.com Sun Dec 27 03:39:17 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 27 Dec 2009 13:39:17 +0200 Subject: [Freeswitch-users] ATA that supports TLS/SRTP w FS In-Reply-To: <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> References: <33c87fa30911212335p1f750411jb4567e232009cf12@mail.gmail.com> <8b1c9cda0912031525v5769ceadv3203652dddb6a0ad@mail.gmail.com> <33c87fa30912031826t30520546x1c02de85fa1fedf8@mail.gmail.com> <33c87fa30912032101y52036e33g772c63d17773a60f@mail.gmail.com> <33c87fa30912162239n35c4a1d1jd74fd43ed628c9c4@mail.gmail.com> Message-ID: More update: VegaStream engineers found the bug and the fix will be available sometime in January. I am still waiting for AudioCodes... Regards, __Yehavi: 2009/12/17 Mark Campbell-Smith > Thanks Yehavi... > > I posted a question on the Cisco Forum and am waiting a response from > 'engineering' to tell us if they plan to implement standard SRTP > support in the Linksys ATA's. > > TLS is working fine. > > On Thu, Dec 17, 2009 at 4:39 PM, Yehavi Bourvine > wrote: > > An interim update: > > > > > > Audiocodes: No success yet. I am working with the manufacturer to debug > it. > > VegaStream: Got the necessary license, configured TLS but it doesn't > work. I > > am working with the local representatives on it. > > > > Regards, __Yehavi: > > > > 2009/12/10 Brian West > >> > >> I have confirmed it works with Polycom, Snom and a few others .... > >> polycom is the hardest to set due to having to put the ca cert into > >> the phone... but other than that its good. > >> > >> /b > >> > >> On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote: > >> > >> > An intermediate report: > >> > > >> > Audiocodes: TLS works only on outgoing requests, incoming ones are > >> > ignored. I am waiting for Audiocodes' help in order to debug it. > >> > SRTP: worked when no TLS is active. When TLS is active the call is > >> > disconnected when the remote party answers. Still debugging it. > >> > > >> > VegaStream Europa-50: SRTP works. Waiting for Vega for instructions > >> > how to enable TLS from the WEB interface. > >> > > >> > Regards, __Yehavi: > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/cdea0c16/attachment-0002.html From kjv at ken-ton.com Sun Dec 27 05:01:03 2009 From: kjv at ken-ton.com (Karl J. Vesterling) Date: Sun, 27 Dec 2009 08:01:03 -0500 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> Message-ID: <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> Setting the codec negotiation to scrooge resolved my problems w/ CallCentric. I'd bet that'd do it for him as well. Lessons Learned by me: 1.) Listen to Brian. 2.) When in doubt, refer to rule 1. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Dec 23, 2009, at 11:57 AM, Brian West wrote: > You might also have to set the codec negotiation to scrooge > > /b > > On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote: > >> You can disable auto-adjust in the sip profile., but that might just make it worse, no warranty: >> >> >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/fce4710f/attachment-0002.html From vizentini at hotmail.com Sun Dec 27 09:14:39 2009 From: vizentini at hotmail.com (Paulo Vicentini) Date: Sun, 27 Dec 2009 17:14:39 +0000 Subject: [Freeswitch-users] SIP registrar Message-ID: Hi,Have you used FS as a registrar server ( handling SIP register/ authorization messages ) with xml_curl for directory lookup?Would FS be suitable for handing registers/authorization messages for about 1,000 U.A.s (with expire 60 s ) ? I am going to make some tests and if you can share your results/experience on this regard it would be very appreciated.I am figuring out if a kamailio/opensips registrar integration with FS is really necessary for a scenario contemplating up to 1K U.A Thanks Paulo _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/4b0d35c8/attachment-0002.html From astmac at stillnewt.org Sun Dec 27 10:52:51 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Sun, 27 Dec 2009 10:52:51 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan Actually, with the release tarballs you don't do bootstrap.sh (unless I am mistaken). I have been building FreeSWITCH on OSX for quite a while (over a year), with good results. I have NOT had any luck building from the SVN, as it seems to throw weird errors on my problems on my platform of choice (PPC OSX Tiger), but the released tarballs seem to work ok (even the pre-release tarballs). I also think that making an OSX package of freeswitch sounds nice, but is a bad idea, UNLESS it's set up in an automated fashion that can stay up to date with changes. Otherwise, lazy OSX people get stuck installing an artifact rather then the best available FreeSWITCH. This happened with Asterisk with the Sunrise telecom people. They ended up creating more problems then good as even years after the fact, silly mac people where still installing the OLD compromised, buggy version just because it was in an OSX installer... Hope this Helps, Marty On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > Ken, > > The process is the same for all UNIX like platforms. You run > > - bootstrap.sh > - configure > - make > - make install > > Jonathan > > On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: > Yes, I want to set up FreeSwitch on OSX and at least see how it > runs, assuming I can get that far. > > I've downloaded the latest tarball and run configure which seemed to > complete ok. But what next? Do I actually need to run > > make all install sounds-install moh-install > > as in some lists of instructions it appears that I just need to run > > make > make install > > Needless to say I'm not an expert at compiling although I have done > a fair bit over the years, just not enough for it to be second > nature. So the above apparent ambiguity puzzles me. > > Also, how can I compile on one machine and then actually run it on a > different machine? Is there a relatively simple way to achieve this > or must I manually copy all the files to the other machine. What > files would that be? Are they all conveniently located in a single > folder? > > Hopeful of some helpful advice, but let's face it, anyone doing this > sort of thing on Christmas Eve really ought to get out more:-) > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gavin.henry at gmail.com Sun Dec 27 16:03:29 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 28 Dec 2009 00:03:29 +0000 Subject: [Freeswitch-users] SIP registrar In-Reply-To: References: Message-ID: <13ca621c0912271603o7a973ce6kbf74ade6fde79f0c@mail.gmail.com> See the installation guide and example dialplans. It's rather simple to test with SIPp and FS. Thanks. On 27/12/2009, Paulo Vicentini wrote: > > Hi,Have you used FS as a registrar server ( handling SIP register/ > authorization messages ) with xml_curl for directory lookup?Would FS be > suitable for handing registers/authorization messages for about 1,000 U.A.s > (with expire 60 s ) ? > I am going to make some tests and if you can share your results/experience > on this regard it would be very appreciated.I am figuring out if a > kamailio/opensips registrar integration with FS is really necessary for a > scenario contemplating up to 1K U.A Thanks > Paulo > _________________________________________________________________ > Windows Live: Make it easier for your friends to see what you?re up to on > Facebook. > http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From sales at cncrepair.com Sun Dec 27 12:22:25 2009 From: sales at cncrepair.com (jim cncrepair.com) Date: Sun, 27 Dec 2009 12:22:25 -0800 Subject: [Freeswitch-users] freeswitch and dahdi give an error: device /dev/zap/channel chan 1 fd 42 (Function not implemented) Message-ID: <4B37C201.3070106@cncrepair.com> I am trying to get freeswitch to talk with a digium tdm400 card. I am using ubuntu and the provided dahdi and freeswitch packages. I have configured /opt/freeswitch/conf/openzap.conf with the following: `--# cat openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 [span zt] name => OpenZAP number => 3 fxo-channel => 3 dahdi's auto generated configuration is the following: `--# cat /etc/dahdi/system.conf # Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" fxsks=1 echocanceller=oslec,1 fxsks=2 echocanceller=oslec,2 fxsks=3 echocanceller=oslec,3 # channel 4, WCTDM/4/3, no module. # Global data loadzone = us defaultzone = us The Freeswitch log is the following: `--# tail -n 900 freeswitch.log| grep zap 2009-12-27 10:05:01.620610 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/modules.conf. 2009-12-27 10:05:01.620738 [NOTICE] zap_io.c:2758 Modules configured: 1 2009-12-27 10:05:01.620765 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/openzap.conf. 2009-12-27 10:05:01.620816 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.621076 [INFO] zap_io.c:2579 Loading IO from /opt/freeswitch/mod/ozmod_zt.so [zt] 2009-12-27 10:05:01.621099 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/zt.conf. 2009-12-27 10:05:01.640363 [INFO] zap_io.c:2379 auto-loaded 'zt' 2009-12-27 10:05:01.651465 [DEBUG] zap_io.c:2400 created span 1 (span1) of type zt 2009-12-27 10:05:01.651607 [DEBUG] zap_io.c:2413 span 1 [name]=[OpenZAP] 2009-12-27 10:05:01.651635 [DEBUG] zap_io.c:2413 span 1 [number]=[1] 2009-12-27 10:05:01.651652 [DEBUG] zap_io.c:2413 span 1 [fxo-channel]=[1] 2009-12-27 10:05:01.651667 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.651847 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 1 fd 42 (Function not implemented) 2009-12-27 10:05:01.651882 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.651968 [DEBUG] zap_io.c:2400 created span 2 (span2) of type zt 2009-12-27 10:05:01.652022 [DEBUG] zap_io.c:2413 span 2 [name]=[OpenZAP] 2009-12-27 10:05:01.652041 [DEBUG] zap_io.c:2413 span 2 [number]=[2] 2009-12-27 10:05:01.652057 [DEBUG] zap_io.c:2413 span 2 [fxo-channel]=[2] 2009-12-27 10:05:01.652071 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.652180 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 2 fd 42 (Function not implemented) 2009-12-27 10:05:01.652207 [DEBUG] zap_io.c:2362 found config for span 2009-12-27 10:05:01.652282 [DEBUG] zap_io.c:2400 created span 3 (span3) of type zt 2009-12-27 10:05:01.652299 [DEBUG] zap_io.c:2413 span 3 [name]=[OpenZAP] 2009-12-27 10:05:01.652315 [DEBUG] zap_io.c:2413 span 3 [number]=[3] 2009-12-27 10:05:01.652344 [DEBUG] zap_io.c:2413 span 3 [fxo-channel]=[3] 2009-12-27 10:05:01.652358 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-27 10:05:01.652483 [ERR] ozmod_zt.c:269 failure configuring device /dev/zap/channel chan 3 fd 42 (Function not implemented) 2009-12-27 10:05:01.652556 [INFO] zap_io.c:2502 Configured 0 channel(s) 2009-12-27 10:05:01.652573 [ERR] zap_io.c:2765 No modules configured! 2009-12-27 10:05:01.652589 [ERR] mod_openzap.c:2882 Error loading OpenZAP 2009-12-27 10:05:01.652605 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_openzap.so To get to this point. I had to symbolically link /dev/zap to /dev/dahdi and change the permissions of /dev/dahdi. I gave /dev/dhadi 777 permissions. I hope someone can advise us what to do next. Thank you From moises.silva at gmail.com Sun Dec 27 20:23:48 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 27 Dec 2009 23:23:48 -0500 Subject: [Freeswitch-users] freeswitch and dahdi give an error: device /dev/zap/channel chan 1 fd 42 (Function not implemented) In-Reply-To: <4B37C201.3070106@cncrepair.com> References: <4B37C201.3070106@cncrepair.com> Message-ID: On Sun, Dec 27, 2009 at 3:22 PM, jim cncrepair.com wrote: > To get to this point. I had to symbolically link /dev/zap to /dev/dahdi > and change the permissions of /dev/dahdi. I gave /dev/dhadi 777 > permissions. > That's just plain wrong. Remove the symlink. -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091227/7a502030/attachment-0002.html From dome at tel.co.th Sun Dec 27 21:07:00 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 28 Dec 2009 12:07:00 +0700 Subject: [Freeswitch-users] What's problem in SVN ? Message-ID: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> Dear sir, What's problem in SVN ? Not thing update after 23/12/2009 (16055) BG Dome C. From scottferri09 at gmail.com Sun Dec 27 22:44:30 2009 From: scottferri09 at gmail.com (Scott Fernandez) Date: Mon, 28 Dec 2009 12:14:30 +0530 Subject: [Freeswitch-users] Cant able to make call through X-lite In-Reply-To: References: Message-ID: Hello , I have a VOIP account which I configured through X-lite and it works fine. However, when I configure the same account in Freeswitch, the status shows as UP. If I call through freeswitch extension (ex. 1001) via X-lite client it says that USER_BUSY. But when I dial the same number through API command, I am able to make a call and bridge it. What could be the issue and Can any one assist me on this? I have pasted the logs in this URL http://pastebin.freeswitch.org/11635 Regards, Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/2fb9025e/attachment-0002.html From darklion11 at yahoo.com Sun Dec 27 23:49:56 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 27 Dec 2009 23:49:56 -0800 (PST) Subject: [Freeswitch-users] Nibblebill not working if mysql reconnects... Message-ID: <26940540.post@talk.nabble.com> Dear Sir, Nibblebill works when i reinstalled and rebuild freeswitch. But after a while when mysql disconnects and reconnects nibblebill accounts not updating recently. IS there another way to avoid nibblebill for not updating? Thanks, Edmar -- View this message in context: http://old.nabble.com/Nibblebill-not-working-if-mysql-reconnects...-tp26940540p26940540.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Dec 28 00:04:07 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 28 Dec 2009 19:04:07 +1100 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> Message-ID: <20091228080407.GA15032@jdc.jasonjgw.net> Dome Charoenyost wrote: > What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. From gmaruzz at celliax.org Mon Dec 28 00:44:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 28 Dec 2009 09:44:36 +0100 Subject: [Freeswitch-users] No gsmopen in trunk? In-Reply-To: References: Message-ID: <7b197bef0912280044o6a11bd70xe28140f9f0304e14@mail.gmail.com> Ciao Max, I'm working on mod_gsmopen, and I can tell you: mod_gsmopen was never in trunk. Actually has not been released yet. So be patient, in the near future I'll announce here on the mailing list how to download a pre-beta of it. -giovanni On Sun, Dec 27, 2009 at 4:51 AM, Max Bridgewater wrote: > Hi, > > I just did a fresh checkout of Freeswitch from trunk but it seems > mod_gsmopen is not there. Am I doing something weird or it has been removed > from trunk. Please advise! > > Thanks, > Max. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dome at tel.co.th Mon Dec 28 05:15:58 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Mon, 28 Dec 2009 20:15:58 +0700 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <20091228080407.GA15032@jdc.jasonjgw.net> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> <20091228080407.GA15032@jdc.jasonjgw.net> Message-ID: <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> Oh...sory.... i forgot chismas and new year. if someone come to thailand please let's me know :) BG Dome C. 2009/12/28 Jason White : > Dome Charoenyost wrote: >> What's problem in SVN ? Not thing update after 23/12/2009 (16055) > > Surely the FreeSWITCH developers are entitled to spend time with their > families/friends after a highly productive year of work. Note that there are > holidays in many countries at this time of year. > > I would like to wish everyone involved in the FreeSWITCH project a pleasant > and refreshing holiday, and much success in 2010. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Mon Dec 28 06:37:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 08:37:43 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> "all" is no longer needed. /b On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > make all install sounds-install moh-install. From brian at freeswitch.org Mon Dec 28 06:46:17 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 08:46:17 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <79116A8B-FCB2-4379-A498-DE53C66C0466@freeswitch.org> You're letting the phones register via the ACL so they never actually do a directory lookup.. domains acl is built from the cidr= attribute on the users in the directory. You have bigger problems if you can't register properly with digest authentication. What does your directory entry look like? /b On Dec 24, 2009, at 2:31 PM, Lars Zeb wrote: > Thanks for the reply, Michael. > > I tried the digest authentication using the cidr and copying the conf/sip_profiles/internal.xml from the distribution, where > > As a result, one endpoint could not register and another was unauthorized. > > http://pastebin.freeswitch.org/11634 > > Then I went changed the context in internal.xml from public to default and > > > And the phones registered OK. So my confusion persists. > > Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/2ecee4e1/attachment-0002.html From brian at freeswitch.org Mon Dec 28 07:25:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 09:25:39 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> Message-ID: <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail. /b On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > This is all I see and then registration fails. From brian at freeswitch.org Mon Dec 28 07:27:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 09:27:15 -0600 Subject: [Freeswitch-users] SNOM shared lines with TLS problems? In-Reply-To: References: Message-ID: <58FEC021-5AA7-484B-B965-D0340ECE968F@freeswitch.org> Shared will require some testing with TLS. I need traces, console logs and you to do some foot work to see if you can provide more details. /b On Dec 24, 2009, at 8:35 AM, Yehavi Bourvine wrote: > Hello, > > Is there anyone who is using SNOM with TLS encryption and shared lines and it works? > > We have 1.0.5pre9 connected to SNOM-820 with shared lines between 2-3 SNOM phones. The TLS is defined by adding transport=tls to the registrar field (proxy is left blank). We noticed the following behaviour: > > With non-shared line UDP and TLS both work ok. > With shared lines UDP works ok. > with shared line TLS works as long as only one phone is registered. > After the second TLS shared line registers we get busy for this extension. From the SNOM trace there is no incoming call attempt at all from FreeSwitch. > Anyone has this setup working and can share some tips? > > Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/c150486a/attachment-0002.html From anthony.minessale at gmail.com Mon Dec 28 07:47:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Dec 2009 09:47:20 -0600 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! In-Reply-To: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> References: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> Message-ID: <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> most likely cause would be connecting a socket then not regularly reading from it causing the buffer to fill up. any event socket connection must select on the socket and do regular read attempts or all the events will accumulate on the server side until some sanity check is reached and it begins to throw them away, the fist time there is room in this buffer again (when you consume some from the socket leaving space in the queue) it will report how many have been lost since the last read. One way to cause this would be suspend fs_cli with ctl-z and bring it back to the foreground after some time. On Thu, Dec 24, 2009 at 7:05 AM, Nicolas Brenner wrote: > I just got into the fs cli and when I ran a 'show calls' I got the > following message: > > 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! > > > What does this mean? does it mean the event_socket did not report 8456 > events? Why could this happen? > > The answer to this is pretty critical to me, as I make and monitor > calls through the socket. > > > Thanks for your help! > > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/625d88a5/attachment-0002.html From jerry.richards at teotech.com Mon Dec 28 08:29:40 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 08:29:40 -0800 Subject: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time In-Reply-To: References: <4B30AB87.3060909@gmx.net> Message-ID: <79F4A32FA1DF436D8BDE8A257B7D3A5A@greyhawk.tonecommander.com> Okay. I uncommented the following lines and the video start works as correctly: Thanks, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time No. The following lines is commented out (internal.xml): Thanks, Jerry -----Original Message----- From: Peter P GMX [mailto:Prometheus001 at gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: > After establishing an audio call between two Bria softphones, and then > starting video at the caller phone, FS replies to the re-INVITE with a > 200 OK with only the PCMU codec. This looks incorrect. The audio > call previously negotiated to the speex/16000 codec, and the re-INVITE > from the caller added the H263-1998 codec. If I re-attempt to start > video at the caller, then it is successful. > > I put a Freeswitch log 11596 into the pastebin that contains the > complete > scenario: establishing audio call, first failed start video attempt, > and second successful start video attempt. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > From darren at aleph-com.net Mon Dec 28 08:39:03 2009 From: darren at aleph-com.net (Darren Wiebe) Date: Mon, 28 Dec 2009 09:39:03 -0700 Subject: [Freeswitch-users] Billing solutions information In-Reply-To: <1261017871151-4179366.post@n2.nabble.com> References: <5d3e0dc60912141111l2a78a89dscbe994b60dc81cbf@mail.gmail.com> <1261017871151-4179366.post@n2.nabble.com> Message-ID: <4B38DF27.2000700@aleph-com.net> On 12/16/2009 7:44 PM, Amarakeerthi S wrote: > Hi, > > Seems nobody is interested to talk about this topic. I found nibblebill is > great. But doesn't hangup the call when balance goes to zero. The other > problem It allows user to call without checking the balance of the cash > database. Is this natural?. If this works fine we can easily integrate with > a payment gateway like 2checkout. > > Thank you > > > > Lon Baker wrote: > >> Hey everyone, >> >> I am researching billing solutions for Freeswitch and want to consolidate >> the information with what others have found, then add it to the Wiki. >> >> There are seems to be a number of billing solutions by commercial >> providers, >> claiming they can integrate with Freeswitch, but nothing concrete >> explaining >> how far they go. >> >> Do they handle processing credit cards, prepaid, postpaid, reporting, lcr, >> etc? >> >> Mod_nibblebill handles the basics of updating a database table. >> >> The A2Billing teams says they are planning on adding support for >> Freeswitch >> in a few months. >> >> ASTPP.org says they support Freeswitch, but the site hasn't been updated >> since 2008. >> >> If you know about any solutions, links to solutions or any information can >> you send it to me? I will organize it and add it to the wiki. >> >> Thanks! >> >> Lon >> >> _______________________________________________ >> The ASTPP site shouldn't show that it hasn't been updated since 2008 as we've been working on it whenever there is time this year. It "works" but needs some more testing and optimizing to be able to handle higher traffic loads. -- Darren Wiebe Aleph Communications Innovative Data& Voice Solutions Email: darren at aleph-com.net Tel: 1-780-701-7267 Fax: 1-866-274-4506 From nicolas at medularis.com Mon Dec 28 09:28:21 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 28 Dec 2009 14:28:21 -0300 Subject: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events! In-Reply-To: <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> References: <1b46b4e80912240505t79d6a2e5l27585e7a3412effd@mail.gmail.com> <191c3a030912280747u5764a2fcw9ed8521ba6a20d1a@mail.gmail.com> Message-ID: <1b46b4e80912280928q30357fb4h27d072c4e8d6f32c@mail.gmail.com> Anthony, thank you very much for your response. The daemon that was reading the events froze, so apparently that was the source of the problem and your explanation fits perfectly. On Mon, Dec 28, 2009 at 12:47 PM, Anthony Minessale wrote: > most likely cause would be connecting a socket then not regularly reading > from it causing the buffer to fill up. > any event socket connection must select on the socket and do regular read > attempts or all the events will accumulate on the server side until some > sanity check is reached and it begins to throw them away, the fist time > there is room in this buffer again (when you consume some from the socket > leaving space in the queue) it will report how many have been lost since the > last read. > > One way to cause this would be suspend fs_cli with ctl-z and bring it back > to the foreground after some time. From mike at jerris.com Mon Dec 28 10:08:04 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Dec 2009 13:08:04 -0500 Subject: [Freeswitch-users] What's problem in SVN ? In-Reply-To: <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> References: <8ccbff060912272107g6d4e0002lf1a151254b1d6e6c@mail.gmail.com> <20091228080407.GA15032@jdc.jasonjgw.net> <8ccbff060912280515hc02b4efse9ac0d3e73909d3a@mail.gmail.com> Message-ID: <6D02BF5B-684B-43C0-92C7-E795A5632124@jerris.com> The issues you ran into are probably sorted out now. Give it a try and if its still not working, post the build errors. Mike On Dec 28, 2009, at 8:15 AM, Dome Charoenyost wrote: > Oh...sory.... i forgot chismas and new year. > if someone come to thailand please let's me know :) > > BG > Dome C. > > > 2009/12/28 Jason White : >> Dome Charoenyost wrote: >>> What's problem in SVN ? Not thing update after 23/12/2009 (16055) >> >> Surely the FreeSWITCH developers are entitled to spend time with their >> families/friends after a highly productive year of work. Note that there are >> holidays in many countries at this time of year. >> >> I would like to wish everyone involved in the FreeSWITCH project a pleasant >> and refreshing holiday, and much success in 2010. >> From jcasale at activenetwerx.com Mon Dec 28 11:54:14 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Mon, 28 Dec 2009 19:54:14 +0000 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: >This is a total work in progress that has not even merged into tree. ?So it is not "known" >to work or not work anywhere. ?Patches to correct issues are welcome. Mike, I took another look at this and don't really know enough about rpm building to diagnose this. Frankly, the format of the latest spec is so wildly different from anything I have ever touched I am at a loss:) Is there a simple manual way for me to properly get the sounds for MOH etc installed? is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location pointed to my /opt/freeswitch/sounds directory? Thanks! jlc From jerry.richards at teotech.com Mon Dec 28 12:21:43 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 12:21:43 -0800 Subject: [Freeswitch-users] Presence Change Distribution Message-ID: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry From mike at jerris.com Mon Dec 28 12:47:47 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 28 Dec 2009 15:47:47 -0500 Subject: [Freeswitch-users] sound rpms In-Reply-To: References: <442A510B-4D9D-484C-A536-998D7066ECDD@jerris.com> Message-ID: the build system already has targets for all of this and there are tarballs you can manually download and extract as well that are located in http://files.freeswitch.org/. if you NEED packages, you will have to wait until that work is complete or figure out what the error is. Mike On Dec 28, 2009, at 2:54 PM, Joseph L. Casale wrote: >> This is a total work in progress that has not even merged into tree. So it is not "known" >> to work or not work anywhere. Patches to correct issues are welcome. > > > Mike, > I took another look at this and don't really know enough about rpm building > to diagnose this. Frankly, the format of the latest spec is so wildly different > from anything I have ever touched I am at a loss:) > > Is there a simple manual way for me to properly get the sounds for MOH etc installed? > is it acceptable to simply run the buildsounds-callie.sh script with the sounds_location > pointed to my /opt/freeswitch/sounds directory? > > Thanks! > jlc > From brian at freeswitch.org Mon Dec 28 13:05:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 15:05:21 -0600 Subject: [Freeswitch-users] twitter.com/freeswitch (its not ours) Message-ID: Dear FreeSWITCHers, Someone has registered the freeswitch name and is squatting on twitter with it. They haven't used it in over a year and I would like to have this for our project as its clearly confusing. If you own this account please contact me off list. Thanks, Brian From msc at freeswitch.org Mon Dec 28 14:25:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 14:25:21 -0800 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <87f2f3b90912281425n2b52579ak5b39d5ac4e97e1ad@mail.gmail.com> On Thu, Dec 24, 2009 at 12:31 PM, Lars Zeb wrote: > Thanks for the reply, Michael. > > > > I tried the digest authentication using the cidr and copying the > conf/sip_profiles/internal.xml from the distribution, where > > > > As a result, one endpoint could not register and another was unauthorized. > > > > http://pastebin.freeswitch.org/11634 > > > > Then I went changed the context in internal.xml from public to default and > > > > > > > And the phones registered OK. So my confusion persists. > > Like Brian said in his post, if you let someone in via ACL then there is no directory lookup which means the call is essentially from an anonymous/unidentified party and thus the reason for having the context set to "public" even on the internal profile. This is one of the topics that I intend to cover in the wiki article. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/cb863847/attachment-0002.html From brian at freeswitch.org Mon Dec 28 14:57:49 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 16:57:49 -0600 Subject: [Freeswitch-users] Local call uses public context? In-Reply-To: <000901ca84d8$1fee8c50$5fcba4f0$@com> References: <012101ca843b$4921cfd0$db656f70$@com> <1151E903-6675-42B2-B32B-4676C06F96AC@freeswitch.org> <002901ca84b4$819f9760$84dec620$@com> <87f2f3b90912241059n2361f5f3nefed7025ad931b0d@mail.gmail.com> <000901ca84d8$1fee8c50$5fcba4f0$@com> Message-ID: <7BAD6694-4FD6-4FEC-A3DF-78BE03129777@freeswitch.org> acl.conf.xml sofia profile: and Then here is an example of a user: Now save that.. restart freeswitch and you now let that user in from 1.2.3.4/32 and set the user_context to default. /b From jerry.richards at teotech.com Mon Dec 28 15:19:41 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 28 Dec 2009 15:19:41 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail Message-ID: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry From anthony.minessale at gmail.com Mon Dec 28 15:30:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Dec 2009 17:30:44 -0600 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail In-Reply-To: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> Message-ID: <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: > Hello All, > > I posted a FS log into the Pastebin at > http://pastebin.freeswitch.org/11644. > > I am still having the problem where a PSTN-to-Internal call via a Sangoma > A101D card stops ringing the internal phone after about 10 seconds. It > should be ringing for 30 seconds and then go to Voice Mail (as an > Internal-to-Internal call does). > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Tuesday, December 22, 2009 8:02 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail > > > I have a Freeswitch PBX server with an installed Sangoma A101D card > connected to a PRI. Most everything works okay, however when I get an > inbound call from the PSTN, if the call is not answered within about 12 > seconds, the call ends (so it doesn't go to voice mail). If I make a call > from one internal phone to another, then it will go to voice mail after 30 > seconds. How can I get the external call to route to voice mail after 30 > seconds? > > I put a new 11595 log into the pastebin. Do you know any Freeswitch > setting > that might cause this? > > If this issue has been addressed before, what string should I use to search > for it, because I can't find it. > > Thanks, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/602e70a4/attachment-0002.html From help at pdscc.com Mon Dec 28 15:38:38 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 15:38:38 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, Message-ID: <20091228233838.75E611694@sinclaire.sibble.net> Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone client, however I am not seeing the enrollment option popup in zfone 0.92 build 218 on windows in front of an x-lite client. Any suggestions on what I should look at to troubleshoot this? I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but until then.... On 14 Dec 2009 at 20:50, Brian West wrote: > if you don't have ZRTP compiled in as per the wiki it won't work... > their are a few changes coming to this code soon. > > /b > > On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote: > > > Hmm, I emailed the zfoneproject folks about an hour ago asking about a > > release date for zfone3 and was surprised about a half hour later > > with a call > > from PRZ himself. > > > > Here's what I got from the call > > > > 1) the currently released version of zfone already has support for > > secure pbx > > enrollment > > -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From brian at freeswitch.org Mon Dec 28 15:49:19 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 28 Dec 2009 17:49:19 -0600 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: <20091228233838.75E611694@sinclaire.sibble.net> References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091215020132.1AD1C1DB501@sinclaire.sibble.net>, <20091228233838.75E611694@sinclaire.sibble.net> Message-ID: I'm still not done with this I think we found a bug in the lib... Viktor fixed it today and I'm going to retry after I get done testing G729 more today! ;) /b On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn trunk, > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone > client, however I am not seeing the enrollment option popup in zfone 0.92 > build 218 on windows in front of an x-lite client. > > Any suggestions on what I should look at to troubleshoot this? > > I am waiting for the Tivi folks to send a 2.0.7 beta for windows mobile, but > until then.... From msc at freeswitch.org Mon Dec 28 16:07:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 16:07:26 -0800 Subject: [Freeswitch-users] forcing ptime settings In-Reply-To: <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> References: <4256bf830912230748g193db15do8e986048836ea5f8@mail.gmail.com> <92E08DEF-19C9-40EA-87CF-CEEEE55B2AA8@freeswitch.org> <4256bf830912230841x3105bacama84514039146e3f0@mail.gmail.com> <3D9F9ABA-841A-4872-998C-58C466F917B5@avgs.ca> <410EFC70-36D0-48A8-BEC9-BBF79050A71D@ken-ton.com> Message-ID: <87f2f3b90912281607w310d8410t44170cf6f20824e0@mail.gmail.com> On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling wrote: > Setting the codec negotiation to scrooge resolved my problems w/ > CallCentric. > > I'd bet that'd do it for him as well. > > *Lessons Learned by me:* > 1.) Listen to Brian. > 2.) When in doubt, refer to rule 1. > Can I get that framed? :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/7aaae93b/attachment-0002.html From msc at freeswitch.org Mon Dec 28 16:09:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Dec 2009 16:09:14 -0800 Subject: [Freeswitch-users] Cant able to make call through X-lite In-Reply-To: References: Message-ID: <87f2f3b90912281609r62b9e25du2bb1783a9acfbb15@mail.gmail.com> On Sun, Dec 27, 2009 at 10:44 PM, Scott Fernandez wrote: > Hello , > > I have a VOIP account which I configured through X-lite and it works fine. > However, when I configure the same account in Freeswitch, the status shows > as UP. If I call through freeswitch extension (ex. 1001) via X-lite client > it says that USER_BUSY. But when I dial the same number through API command, > I am able to make a call and bridge it. > > What could be the issue and Can any one assist me on this? > > I have pasted the logs in this URL http://pastebin.freeswitch.org/11635 > > This pb no more... can you re-post? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091228/8e56859b/attachment-0002.html From help at pdscc.com Mon Dec 28 16:39:26 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 16:39:26 -0800 Subject: [Freeswitch-users] zrtp endpoints have different sas through fs 1.0.4 In-Reply-To: References: <20090823213745.D81819D5@sinclaire.sibble.net>, <20091228233838.75E611694@sinclaire.sibble.net>, Message-ID: <20091229003926.072E513F5@sinclaire.sibble.net> Coolio, well if you need me to test something, just holler. here's what I'm running FreeSWITCH Version 1.0.trunk (16066) on Ubuntu 9.0.4 zfone 0.92 build 218 (windows xp) with ekiga and x-lite clients tiviphone 2.0.7 (beta) for symbian hopefully will have the tiviphone 2.0.7 (beta) for windows mobile shortly On 28 Dec 2009 at 17:49, Brian West wrote: > I'm still not done with this I think we found a bug in the lib... Viktor > fixed it today and I'm going to retry after I get done testing G729 more > today! ;) > > /b > > On Dec 28, 2009, at 5:38 PM, Harondel J. Sibble wrote: > > > Hmm, okay, I went back to basics and did a full rebuild for 1.0.4 svn > trunk, > > I am able to now enroll my nokia e61i running the beta 2.0.7 Tiviphone > > client, however I am not seeing the enrollment option popup in zfone 0.92 > > build 218 on windows in front of an x-lite client. > > > > Any suggestions on what I should look at to troubleshoot this? -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From help at pdscc.com Mon Dec 28 21:21:06 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Mon, 28 Dec 2009 21:21:06 -0800 Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch Message-ID: <20091229052106.2592813F5@sinclaire.sibble.net> I am following the wiki page here http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation setup is on Ubuntu 9.0.4 using the debian method, when I run module-assistant build zaptel-source the compilation fails as below, not sure what I am missing, reading further FS wiki pages and googling haven't enlightened me, any suggestions? I am trying to get 2x X100P's working dh_testdir dh_testroot rm -f *-stamp # Delete the generated bristuff symlinks: rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch] # Add here commands to clean up after the build process. rm -rf modexamples rm -f tonezones.txt rm -f version.h rm -rf debian/fake # * Makefile does not exist when running svn-buildpackage # as the source tree is not there. # FIXME: This will fail with an ugly warning on the clean of the # modules build. However only fter the actuual clean. [ ! -f Makefile ] || /usr/bin/make dist-clean || true make[1]: Entering directory `/usr/src/modules/zaptel' make: Entering an unknown directory make: Leaving an unknown directory rm -f torisatool rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest ztscan rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo /usr/bin/make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor make[2]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' CLEAN /usr/src/modules/zaptel/kernel CLEAN /usr/src/modules/zaptel/kernel/.tmp_versions make[2]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[2]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils' rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8 lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8 make[2]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils' make: Entering an unknown directory make: Leaving an unknown directory make[1]: Leaving directory `/usr/src/modules/zaptel' #rm -f debian/manpage.links debian/manpage.refs debian/*.8 dh_clean /usr/bin/make -f debian/rules kdist_clean kdist_config binary-modules make[1]: Entering directory `/usr/src/modules/zaptel' dh_testdir dh_testroot rm -f *-stamp # Delete the generated bristuff symlinks: rm -f -f cwain.[ch] qozap.[ch] zaphfc.[ch] ztgsm.[ch] # Add here commands to clean up after the build process. rm -rf modexamples rm -f tonezones.txt rm -f version.h rm -rf debian/fake # * Makefile does not exist when running svn-buildpackage # as the source tree is not there. # FIXME: This will fail with an ugly warning on the clean of the # modules build. However only fter the actuual clean. [ ! -f Makefile ] || /usr/bin/make dist-clean || true make[2]: Entering directory `/usr/src/modules/zaptel' make: Entering an unknown directory make: *** menuselect: No such file or directory. Stop. make: Leaving an unknown directory make[2]: [clean] Error 2 (ignored) rm -f torisatool rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest ztscan rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo /usr/bin/make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor make[3]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' make[3]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[3]: Entering directory `/usr/src/modules/zaptel/kernel/xpp/utils' rm -f *.o init_fxo_modes print_modes perlcheck zt_registration.8 xpp_sync.8 lszaptel.8 xpp_blink.8 zapconf.8 zaptel_hardware.8 make[3]: Leaving directory `/usr/src/modules/zaptel/kernel/xpp/utils' make: Entering an unknown directory make: *** ppp: No such file or directory. Stop. make: Leaving an unknown directory make[2]: *** [clean] Error 2 make[2]: Leaving directory `/usr/src/modules/zaptel' #rm -f debian/manpage.links debian/manpage.refs debian/*.8 dh_clean for templ in ; do \ cp $templ `echo $templ | sed -e 's/_KVERS_/2.6.28-11-server/g'` ; \ done for templ in `ls debian/*.modules.in` ; do \ test -e ${templ%.modules.in}.backup || cp ${templ%.modules.in} ${templ%.modules.in}.backup 2>/dev/null || true; \ sed -e 's/##KVERS##/2.6.28-11-server/g ;s/#KVERS#/2.6.28-11-server/g ; s/_KVERS_/2.6.28-11-server/g ; s/##KDREV##/2.6.28-11.42 done dh_testdir dh_testroot dh_clean -k cp -a /usr/src/modules/zaptel/debian/generated/* . ./configure checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/bash checking for ln... /bin/ln checking for wget... /usr/bin/wget checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... no checking for usb_init in -lusb... no configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts config.status: creating build_tools/make_firmware_object configure: *** Zaptel build successfully configured *** make MODULES_EXTRA="ds1x1f opvxa1200 wcopenpci cwain qozap zaphfc ztgsm" SUBDIRS_EXTRA="vzaphfc oslec" modules KERNEL_SOURCES=/usr make[2]: Entering directory `/usr/src/modules/zaptel' make -C /usr/src/linux ARCH=i386 SUBDIRS=/usr/src/modules/zaptel/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor2.o toris make[3]: Entering directory `/usr/src/linux-headers-2.6.28-11-server' gcc -o /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/makefw.c /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/pciradio.rbt radfw > /usr/src/modules/zaptel/kernel/radfw.h Loaded 42096 bytes from file CC [M] /usr/src/modules/zaptel/kernel/pciradio.o /usr/src/modules/zaptel/kernel/makefw /usr/src/modules/zaptel/kernel/tormenta2.rbt tor2fw > /usr/src/modules/zaptel/kernel/tor2fw. Loaded 69900 bytes from file CC [M] /usr/src/modules/zaptel/kernel/tor2.o CC [M] /usr/src/modules/zaptel/kernel/torisa.o CC [M] /usr/src/modules/zaptel/kernel/wcfxo.o CC [M] /usr/src/modules/zaptel/kernel/wct1xxp.o CC [M] /usr/src/modules/zaptel/kernel/wctdm.o /usr/src/modules/zaptel/kernel/wctdm.c: In function 'wctdm_proslic_getreg_indirect': /usr/src/modules/zaptel/kernel/wctdm.c:671: warning: format not a string literal and no format arguments CC [M] /usr/src/modules/zaptel/kernel/wcte11xp.o CC [M] /usr/src/modules/zaptel/kernel/wcusb.o CC [M] /usr/src/modules/zaptel/kernel/zaptel-base.o /usr/src/modules/zaptel/kernel/zaptel-base.c: In function 'zt_ppp_xmit': /usr/src/modules/zaptel/kernel/zaptel-base.c:1751: warning: comparison of distinct pointer types lacks a cast /usr/src/modules/zaptel/kernel/zaptel-base.c:1814: warning: comparison of distinct pointer types lacks a cast LD [M] /usr/src/modules/zaptel/kernel/zaptel.o CC [M] /usr/src/modules/zaptel/kernel/ztd-eth.o CC [M] /usr/src/modules/zaptel/kernel/ztd-loc.o CC [M] /usr/src/modules/zaptel/kernel/ztdummy.o /usr/src/modules/zaptel/kernel/ztdummy.c: In function 'ztdummy_hr_int': /usr/src/modules/zaptel/kernel/ztdummy.c:203: error: 'struct hrtimer' has no member named 'expires' make[4]: *** [/usr/src/modules/zaptel/kernel/ztdummy.o] Error 1 make[3]: *** [_module_/usr/src/modules/zaptel/kernel] Error 2 make[3]: Leaving directory `/usr/src/linux-headers-2.6.28-11-server' make[2]: *** [modules] Error 2 make[2]: Leaving directory `/usr/src/modules/zaptel' make[1]: *** [binary-modules] Error 2 make[1]: Leaving directory `/usr/src/modules/zaptel' make: *** [kdist_build] Error 2 -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From sharad at coraltele.com Mon Dec 28 22:00:43 2009 From: sharad at coraltele.com (Sharad) Date: Mon, 28 Dec 2009 22:00:43 -0800 (PST) Subject: [Freeswitch-users] Personal Greeting Message-ID: <1262066443847-4226681.post@n2.nabble.com> Hi I am new to Freeswitch so my question may be a stupid question. I just want to know how to disable the personal greeting to the default one. One user has recorded his personal greeting now how can he make this default. I could not find any option for the same. Plz advice. Thanks & regards Sharad garg -- View this message in context: http://n2.nabble.com/Personal-Greeting-tp4226681p4226681.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sharad at coraltele.com Mon Dec 28 22:23:20 2009 From: sharad at coraltele.com (Sharad) Date: Mon, 28 Dec 2009 22:23:20 -0800 (PST) Subject: [Freeswitch-users] Message Wait Lamp for an unregistered user Message-ID: <1262067800887-4226726.post@n2.nabble.com> Hi, I am trying to integrate Freeswitch as a Media Server with our own SIP Server. For this, initially, I am using freeswitch for Auto Attendant & Voicemail Application. In this case, all the SIP users are registered with my SIP Server & whenever Auto Att or Voicemail application is required, my SIP Server just forward the call to Freeswitch. Everything seems ok & working fine except some small small points. One of the point is - whenever there is a voice message in the mailbox of a user, Freeswitch is not generating the MWI (Notify) to my SIP Server. So just want to know , is there any way so that freeswitch can light up / light -off the message wait lamp for the users of my SIP Server. Thanks in advance for your answers. Regards Sharad -- View this message in context: http://n2.nabble.com/Message-Wait-Lamp-for-an-unregistered-user-tp4226726p4226726.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lei.tlfly at gmail.com Mon Dec 28 23:14:51 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 15:14:51 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 Message-ID: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. The problem is, when send reponse for re-invite request, fs didn't send any sdp content. This problem is easy to reproduce, just call to fs, and press hold button, Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. ====sip trace for trunk 16055 ====re-invite request sent to fs when client hold the line INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-Id: s264bdfe05129544c7e0a2c44408cb213 Cseq: 12860 INVITE Contact: > Content-Type: application/sdp Content-Length: 462 Date: Tue, 29 Dec 2009 06:53:53 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport v=0 o=sipX 5 6 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE User-Agent: PowerIVR Content-Length: 0 =====bad response sent by fs, sdp content is missing. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 From: ;tag=1c6494 To: ;tag=tUS6Q8KmtmDZe Call-ID: s264bdfe05129544c7e0a2c44408cb213 CSeq: 12860 INVITE Contact: User-Agent: PowerIVR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Length: 0 ======sip trace for fs 1.0.4 =====re-invite request sent to FS when client want to hold the all INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 INVITE Contact: > Content-Type: application/sdp Content-Length: 463 Date: Tue, 29 Dec 2009 03:20:14 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport v=0 o=sipX 5 34 IN IP4 0.0.0.0 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE User-Agent: PowerIVR Content-Length: 0 ===repsonse sent by fs, there is correct sdp content. SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29657 INVITE Contact: User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 254 v=0 o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 Contact: > From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29657 ACK Date: Tue, 29 Dec 2009 03:20:15 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport Content-Length: 0 INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 INVITE Contact: > Content-Type: application/sdp Content-Length: 473 Date: Tue, 29 Dec 2009 03:20:18 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY Supported: replaces Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport v=0 o=sipX 5 35 IN IP4 10.56.90.223 s=call c=IN IP4 10.56.90.223 t=0 0 m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=3 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=2 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=5 a=rtpmap:113 speex/8000/1 a=fmtp:113 mode=7 a=rtpmap:3 gsm/8000/1 a=rtpmap:97 ilbc/8000/1 a=fmtp:97 mode=30 a=ptime:30 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE User-Agent: PowerIVR Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-ID: s8fc27f8446522ddd375f0e20d43e5aad CSeq: 29658 INVITE Contact: User-Agent: PowerIVR Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 242 v=0 o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 s=FreeSWITCH c=IN IP4 10.56.0.189 t=0 0 m=audio 28606 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 Contact: > From: ;tag=1c8147 To: ;tag=tH78Sc30vXKXK Call-Id: s8fc27f8446522ddd375f0e20d43e5aad Cseq: 29658 ACK Date: Tue, 29 Dec 2009 03:20:19 GMT Max-Forwards: 70 User-Agent: SipPhone Accept-Language: en Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport Content-Length: 0 -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/4e5eb8a2/attachment-0002.html From siniypin at gmail.com Tue Dec 29 00:02:24 2009 From: siniypin at gmail.com (RobertT) Date: Tue, 29 Dec 2009 11:02:24 +0300 Subject: [Freeswitch-users] Presence Change Distribution In-Reply-To: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> References: <96888A19920E403880AAA9F6EE061BB6@greyhawk.tonecommander.com> Message-ID: <2160023e0912290002q3d0f3fden5adee6d87d4bde25@mail.gmail.com> You can try to reduce your registration time. I for one made my client apps send PUBLISH message every minute in addition to reduced registration time. Regards, Robert. 2009/12/28 Jerry Richards > Is there a setting to control how fast FS distributes presence changes to > subscribers? Currently, it appears to take several minutes before I see > presence changes. I would like to see them almost instantaneously, if > possible. > > Thanks and Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/94caa383/attachment-0002.html From Russell.Mosemann at cune.org Tue Dec 29 04:14:38 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Tue, 29 Dec 2009 06:14:38 -0600 Subject: [Freeswitch-users] problems getting openzap compiled for use withfreeswitch In-Reply-To: <20091229052106.2592813F5@sinclaire.sibble.net> References: <20091229052106.2592813F5@sinclaire.sibble.net> Message-ID: <291960A5CE644885B77540F79638F0AC@cune.pri> Harondel J. Sibble scribbled: > I am following the wiki page here > > http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation ... > the compilation fails as below, Zaptel is old. DAHDI is now the way to go. If you have the headers installed for the kernel you are currently using, try this. 1. http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz 2. tar -xzf dahdi-linux-complete-current.tar.gz 3. cd dahdi-linux-complete- 4. make all 5. make install 6. make config (First time only! Modify config files in /etc/dahdi) If this works, please add a DAHDI section to the wiki. If you want to use a Debian package instead of compiling the latest from scratch, try the dahdi-linux package. -- Russell Mosemann From freeswitch-list at puzzled.xs4all.nl Tue Dec 29 06:09:31 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Tue, 29 Dec 2009 15:09:31 +0100 Subject: [Freeswitch-users] problems getting openzap compiled for use withfreeswitch In-Reply-To: <291960A5CE644885B77540F79638F0AC@cune.pri> References: <20091229052106.2592813F5@sinclaire.sibble.net> <291960A5CE644885B77540F79638F0AC@cune.pri> Message-ID: <4B3A0D9B.7030708@puzzled.xs4all.nl> On 12/29/2009 01:14 PM, Russell Mosemann wrote: > Harondel J. Sibble scribbled: >> I am following the wiki page here >> >> http://wiki.freeswitch.org/wiki/OpenZAP#Zaptel_Installation > ... >> the compilation fails as below, > > Zaptel is old. DAHDI is now the way to go. If you have the headers installed for the kernel you are currently using, try this. > > 1. http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz > 2. tar -xzf dahdi-linux-complete-current.tar.gz > 3. cd dahdi-linux-complete- > 4. make all > 5. make install > 6. make config (First time only! Modify config files in /etc/dahdi) > > If this works, please add a DAHDI section to the wiki. If you want to use a Debian package instead of compiling the latest from scratch, try the dahdi-linux package. Or you could ask Tzafrir in #asterisk where the latest and greatest dahdi deb packages are as he is afaik the maintainer. Regards, Patrick From brian at freeswitch.org Tue Dec 29 06:35:12 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 08:35:12 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: Hold is working fine I just tested it... I would need to see the whole dialog to see what is wrong... I tested with Polycom, Snom and Aastra. Are you doing proxy media or anything like that? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/49e690c1/attachment-0002.html From brian at freeswitch.org Tue Dec 29 06:51:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 08:51:36 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> Also can you join #freeswitch-dev, include full siptrace+debug log and put it on pastebin. What phone are you using? /b On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > Hi, I think hold function in trunk 16055 is broken, I have also tried some old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/5412677f/attachment-0002.html From lei.tlfly at gmail.com Tue Dec 29 06:53:41 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 22:53:41 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> Message-ID: <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. I will test if trunk 16055 work when I set proxy media mode to false tomorrow. 2009/12/29 Brian West > Hold is working fine I just tested it... I would need to see the whole > dialog to see what is wrong... I tested with Polycom, Snom and Aastra. > > Are you doing proxy media or anything like that? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/7c7246a0/attachment-0002.html From lei.tlfly at gmail.com Tue Dec 29 07:10:59 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:10:59 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> Message-ID: <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> The phone I'm using is x-lite and wxCommunicator, both are sip phone software. I have not used pastebin, Is it a bug trace tool like bugzilla? Can you tell me how to register a pastbin account? 2009/12/29 Brian West > Also can you join #freeswitch-dev, include full siptrace+debug log and put > it on pastebin. > > What phone are you using? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > > Hi, I think hold function in trunk 16055 is broken, I have also tried some > old trunks, it's ok in freeswitch 1.0.4. > The problem is, when send reponse for re-invite request, fs didn't send any > sdp content. > This problem is easy to reproduce, just call to fs, and press hold button, > Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are both > included. > > ====sip trace for trunk 16055 > ====re-invite request sent to fs when client hold the line > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-Id: s264bdfe05129544c7e0a2c44408cb213 > Cseq: 12860 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 462 > Date: Tue, 29 Dec 2009 06:53:53 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport > > v=0 > o=sipX 5 6 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > =====bad response sent by fs, sdp content is missing. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 > From: ;tag=1c6494 > To: ;tag=tUS6Q8KmtmDZe > Call-ID: s264bdfe05129544c7e0a2c44408cb213 > CSeq: 12860 INVITE > Contact: > User-Agent: PowerIVR > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Length: 0 > > > ======sip trace for fs 1.0.4 > =====re-invite request sent to FS when client want to hold the all > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 463 > Date: Tue, 29 Dec 2009 03:20:14 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > > v=0 > o=sipX 5 34 IN IP4 0.0.0.0 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > ===repsonse sent by fs, there is correct sdp content. > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29657 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 254 > > v=0 > o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=recvonly > a=silenceSupp:off - - - - > a=ptime:20 > > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29657 ACK > Date: Tue, 29 Dec 2009 03:20:15 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport > Content-Length: 0 > > INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 INVITE > Contact: > > Content-Type: application/sdp > Content-Length: 473 > Date: Tue, 29 Dec 2009 03:20:18 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, MESSAGE, > REGISTER, NOTIFY > Supported: replaces > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > > v=0 > o=sipX 5 35 IN IP4 10.56.90.223 > s=call > c=IN IP4 10.56.90.223 > t=0 0 > m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 > a=rtpmap:0 pcmu/8000/1 > a=rtpmap:8 pcma/8000/1 > a=rtpmap:96 telephone-event/8000/1 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=3 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=2 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=5 > a=rtpmap:113 speex/8000/1 > a=fmtp:113 mode=7 > a=rtpmap:3 gsm/8000/1 > a=rtpmap:97 ilbc/8000/1 > a=fmtp:97 mode=30 > a=ptime:30 > > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > User-Agent: PowerIVR > Content-Length: 0 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-ID: s8fc27f8446522ddd375f0e20d43e5aad > CSeq: 29658 INVITE > Contact: > User-Agent: PowerIVR > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Session-Expires: 120;refresher=uas > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 242 > > v=0 > o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 > s=FreeSWITCH > c=IN IP4 10.56.0.189 > t=0 0 > m=audio 28606 RTP/AVP 8 96 > a=rtpmap:8 pcma/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 > Contact: > > From: ;tag=1c8147 > To: ;tag=tH78Sc30vXKXK > Call-Id: s8fc27f8446522ddd375f0e20d43e5aad > Cseq: 29658 ACK > Date: Tue, 29 Dec 2009 03:20:19 GMT > Max-Forwards: 70 > User-Agent: SipPhone > Accept-Language: en > Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport > Content-Length: 0 > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/59271f9c/attachment-0002.html From brian at freeswitch.org Tue Dec 29 07:14:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 09:14:22 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> Message-ID: <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> the 200ok is not from FS.. its from the end point... so its not us thats not putting the SDP into the 200ok but the device you're talking to because in proxy media they are passed as is. /b On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > Hi Brian, thanks for your help, I am using FS in proxy media mode. the sip agent I'm using is x-lite and wxCommunicator. > I will test if trunk 16055 work when I set proxy media mode to false tomorrow. From lei.tlfly at gmail.com Tue Dec 29 07:37:22 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:37:22 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> Message-ID: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following code in sofia.c send the 200ok response sofia.c function sofia_handle_sip_i_state .... ......... switch(ss_state) ................ case nua_callstate_received: ..................... else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && !r_sdp) { nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); goto done; } The cause is r_sdp is null, but I don't known why tl_gets don't return remote sdp tag, it's quite strange. 2009/12/29 Brian West > the 200ok is not from FS.. its from the end point... so its not us thats > not putting the SDP into the 200ok but the device you're talking to because > in proxy media they are passed as is. > > /b > > On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > > > Hi Brian, thanks for your help, I am using FS in proxy media mode. the > sip agent I'm using is x-lite and wxCommunicator. > > I will test if trunk 16055 work when I set proxy media mode to false > tomorrow. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/1d4c23f7/attachment-0002.html From lei.tlfly at gmail.com Tue Dec 29 07:38:35 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Tue, 29 Dec 2009 23:38:35 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <50c41b4e0912290738x73d10086kcbf40973c092f2ca@mail.gmail.com> Btw, in the same scenario, FS 1.0.4 works fine. 2009/12/29 Lei Tang > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following > code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && > !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return > remote sdp tag, it's quite strange. > > 2009/12/29 Brian West > >> the 200ok is not from FS.. its from the end point... so its not us thats >> not putting the SDP into the 200ok but the device you're talking to because >> in proxy media they are passed as is. >> >> >> /b >> >> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >> >> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >> sip agent I'm using is x-lite and wxCommunicator. >> > I will test if trunk 16055 work when I set proxy media mode to false >> tomorrow. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/1bc15a3c/attachment-0002.html From mike at jerris.com Tue Dec 29 07:39:14 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 10:39:14 -0500 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <812A0608-1097-4F44-A2BD-FB3D801D35BA@freeswitch.org> <50c41b4e0912290710w3f3719eaq9ffe510ca864c058@mail.gmail.com> Message-ID: <2BE60334-889A-44B9-99DE-AD1EDEBEB816@jerris.com> There is no need for you to show us traces. The fact that you are using proxy media is enough to know that the issue is with your device. If you look at the full sip trace you will see the same. Mike On Dec 29, 2009, at 10:10 AM, Lei Tang wrote: > The phone I'm using is x-lite and wxCommunicator, both are sip > phone software. > I have not used pastebin, Is it a bug trace tool like bugzilla? Can > you tell me how to register a pastbin account? > > 2009/12/29 Brian West > Also can you join #freeswitch-dev, include full siptrace+debug log > and put it on pastebin. > > What phone are you using? > > /b > > On Dec 29, 2009, at 1:14 AM, Lei Tang wrote: > >> Hi, I think hold function in trunk 16055 is broken, I have also >> tried some old trunks, it's ok in freeswitch 1.0.4. >> The problem is, when send reponse for re-invite request, fs didn't >> send any sdp content. >> This problem is easy to reproduce, just call to fs, and press hold >> button, >> Follow are sip trace messages I catched, trunk 16055 and 1.0.4 are >> both included. >> >> ====sip trace for trunk 16055 >> ====re-invite request sent to fs when client hold the line >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-Id: s264bdfe05129544c7e0a2c44408cb213 >> Cseq: 12860 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 462 >> Date: Tue, 29 Dec 2009 06:53:53 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport >> >> v=0 >> o=sipX 5 6 IN IP4 0.0.0.0 >> s=call >> c=IN IP4 0.0.0.0 >> t=0 0 >> m=audio 9000 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-ID: s264bdfe05129544c7e0a2c44408cb213 >> CSeq: 12860 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> =====bad response sent by fs, sdp content is missing. >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bad9a41468c5;rport=5060 >> From: ;tag=1c6494 >> To: ;tag=tUS6Q8KmtmDZe >> Call-ID: s264bdfe05129544c7e0a2c44408cb213 >> CSeq: 12860 INVITE >> Contact: >> User-Agent: PowerIVR >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Length: 0 >> >> >> ======sip trace for fs 1.0.4 >> =====re-invite request sent to FS when client want to hold the all >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29657 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 463 >> Date: Tue, 29 Dec 2009 03:20:14 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport >> >> v=0 >> o=sipX 5 34 IN IP4 0.0.0.0 >> s=call >> c=IN IP4 0.0.0.0 >> t=0 0 >> m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29657 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> ===repsonse sent by fs, there is correct sdp content. >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29657 INVITE >> Contact: >> User-Agent: PowerIVR >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 254 >> >> v=0 >> o=FreeSWITCH 1262028193 1262028195 IN IP4 10.56.0.189 >> s=FreeSWITCH >> c=IN IP4 10.56.0.189 >> t=0 0 >> m=audio 28606 RTP/AVP 8 96 >> a=rtpmap:8 pcma/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> a=recvonly >> a=silenceSupp:off - - - - >> a=ptime:20 >> >> ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> Contact: >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29657 ACK >> Date: Tue, 29 Dec 2009 03:20:15 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-bbdf6f9a536f;rport >> Content-Length: 0 >> >> INVITE sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29658 INVITE >> Contact: >> Content-Type: application/sdp >> Content-Length: 473 >> Date: Tue, 29 Dec 2009 03:20:18 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PING, INFO, >> MESSAGE, REGISTER, NOTIFY >> Supported: replaces >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport >> >> v=0 >> o=sipX 5 35 IN IP4 10.56.90.223 >> s=call >> c=IN IP4 10.56.90.223 >> t=0 0 >> m=audio 9002 RTP/AVP 0 8 96 113 113 113 113 3 97 >> a=rtpmap:0 pcmu/8000/1 >> a=rtpmap:8 pcma/8000/1 >> a=rtpmap:96 telephone-event/8000/1 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=3 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=2 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=5 >> a=rtpmap:113 speex/8000/1 >> a=fmtp:113 mode=7 >> a=rtpmap:3 gsm/8000/1 >> a=rtpmap:97 ilbc/8000/1 >> a=fmtp:97 mode=30 >> a=ptime:30 >> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29658 INVITE >> User-Agent: PowerIVR >> Content-Length: 0 >> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport=5060 >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-ID: s8fc27f8446522ddd375f0e20d43e5aad >> CSeq: 29658 INVITE >> Contact: >> User-Agent: PowerIVR >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH >> Supported: timer, precondition, path, replaces >> Session-Expires: 120;refresher=uas >> Min-SE: 120 >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 242 >> >> v=0 >> o=FreeSWITCH 1262028193 1262028196 IN IP4 10.56.0.189 >> s=FreeSWITCH >> c=IN IP4 10.56.0.189 >> t=0 0 >> m=audio 28606 RTP/AVP 8 96 >> a=rtpmap:8 pcma/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> ACK sip:65960581 at 10.56.0.189:5060;transport=udp SIP/2.0 >> Contact: >> From: ;tag=1c8147 >> To: ;tag=tH78Sc30vXKXK >> Call-Id: s8fc27f8446522ddd375f0e20d43e5aad >> Cseq: 29658 ACK >> Date: Tue, 29 Dec 2009 03:20:19 GMT >> Max-Forwards: 70 >> User-Agent: SipPhone >> Accept-Language: en >> Via: SIP/2.0/UDP 10.56.90.223;branch=z9hG4bK-01413ea87577;rport >> Content-Length: 0 >> >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/d25b0838/attachment-0002.html From brian at freeswitch.org Tue Dec 29 07:42:57 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 09:42:57 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <428218D4-D5C8-4AF3-B8F8-72182935CC41@freeswitch.org> Its null because the device on the other side didn't send one. We pass it as is... fix the broken device or don't use proxy media. /b On Dec 29, 2009, at 9:37 AM, Lei Tang wrote: > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return remote sdp tag, it's quite strange. From mike at jerris.com Tue Dec 29 08:08:08 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 11:08:08 -0500 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> Message-ID: <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> This means there was no sdp sent. Did you confirm this with siptrace? On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, > following code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, > TFLAG_SDP) && !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, > TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't > return remote sdp tag, it's quite strange. > > 2009/12/29 Brian West > the 200ok is not from FS.. its from the end point... so its not us > thats not putting the SDP into the 200ok but the device you're > talking to because in proxy media they are passed as is. > > /b > > On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: > > > Hi Brian, thanks for your help, I am using FS in proxy media mode. > the sip agent I'm using is x-lite and wxCommunicator. > > I will test if trunk 16055 work when I set proxy media mode to > false tomorrow. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > -- > Lei.Tang > lei.tlfly at gmail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/a6d170bc/attachment-0002.html From anthony.minessale at gmail.com Tue Dec 29 08:45:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Dec 2009 10:45:18 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> Message-ID: <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> We now disable sofia SOA mode during proxy calls. This means that sofia will not try to get involved in the media negotiation at all which is the optimal behavior. Previous versions would butt in and try to fix the error but now it just stays out of the way. You can see in your trace that the device sends a packet with no SDP therefore so does sofia. You can either turn off proxy-media or post a bounty for me to go hack a workaround into the patch I spent many hours on getting things to work right. Whatever you experienced with 1.0.4 was a happy coincidence where sofia was fixing a bug in your phone for you. On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: > This means there was no sdp sent. Did you confirm this with siptrace? > > On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: > > Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following > code in sofia.c send the 200ok response > sofia.c > function sofia_handle_sip_i_state .... > ......... > switch(ss_state) > ................ > case nua_callstate_received: > ..................... > else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && > !r_sdp) { > nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); > sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); > goto done; > } > > The cause is r_sdp is null, but I don't known why tl_gets don't return > remote sdp tag, it's quite strange. > > 2009/12/29 Brian West < brian at freeswitch.org> > >> the 200ok is not from FS.. its from the end point... so its not us thats >> not putting the SDP into the 200ok but the device you're talking to because >> in proxy media they are passed as is. >> >> /b >> >> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >> >> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >> sip agent I'm using is x-lite and wxCommunicator. >> > I will test if trunk 16055 work when I set proxy media mode to false >> tomorrow. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/48a309e1/attachment-0002.html From ivan at myrvold.org Tue Dec 29 09:06:45 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 18:06:45 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. Ivan Den 27. des. 2009 kl. 19.52 skrev Martin Joseph: > On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > >> Ken, >> >> The process is the same for all UNIX like platforms. You run >> >> - bootstrap.sh >> - configure >> - make >> - make install >> >> Jonathan > > > > > Actually, with the release tarballs you don't do bootstrap.sh (unless > I am mistaken). > > I have been building FreeSWITCH on OSX for quite a while (over a > year), with good results. I have NOT had any luck building from the > SVN, as it seems to throw weird errors on my problems on my platform > of choice (PPC OSX Tiger), but the released tarballs seem to work ok > (even the pre-release tarballs). > > I also think that making an OSX package of freeswitch sounds nice, > but is a bad idea, UNLESS it's set up in an automated fashion that can > stay up to date with changes. Otherwise, lazy OSX people get stuck > installing an artifact rather then the best available FreeSWITCH. This > happened with Asterisk with the Sunrise telecom people. They ended up > creating more problems then good as even years after the fact, silly > mac people where still installing the OLD compromised, buggy version > just because it was in an OSX installer... > > Hope this Helps, > Marty > > On Dec 24, 2009, at 8:40 PM, jonathan augenstine wrote: > >> Ken, >> >> The process is the same for all UNIX like platforms. You run >> >> - bootstrap.sh >> - configure >> - make >> - make install >> >> Jonathan >> >> On Thu, Dec 24, 2009 at 12:42 PM, Ken Gillett wrote: >> Yes, I want to set up FreeSwitch on OSX and at least see how it >> runs, assuming I can get that far. >> >> I've downloaded the latest tarball and run configure which seemed to >> complete ok. But what next? Do I actually need to run >> >> make all install sounds-install moh-install >> >> as in some lists of instructions it appears that I just need to run >> >> make >> make install >> >> Needless to say I'm not an expert at compiling although I have done >> a fair bit over the years, just not enough for it to be second >> nature. So the above apparent ambiguity puzzles me. >> >> Also, how can I compile on one machine and then actually run it on a >> different machine? Is there a relatively simple way to achieve this >> or must I manually copy all the files to the other machine. What >> files would that be? Are they all conveniently located in a single >> folder? >> >> Hopeful of some helpful advice, but let's face it, anyone doing this >> sort of thing on Christmas Eve really ought to get out more:-) >> >> >> >> Ken G i l l e t t >> >> _/_/_/_/_/_/_/_/ >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 09:11:40 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 11:11:40 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> Message-ID: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Ivan, I have been trying to gather up everyone to start a FreeSWITCH based softphone project for Mac, Linux and Windows... you think we could collaborate with you to accomplish this? I think if we do this right we can have a really nice phone with lots of options. Thanks, /b On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. > I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. > > Ivan > From mrene_lists at avgs.ca Tue Dec 29 10:14:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 29 Dec 2009 19:14:37 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <8DA219E3-A980-418E-A147-6AA26788E0BB@avgs.ca> This could be easily done with the Qt framework and would work nicely on osx, linux and windows. Contact me off list or on IRC (Math) if you want some help, I'd be happy to participate in such a project. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 29-Dec-09, at 6:11 PM, Brian West wrote: > Ivan, > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we > could collaborate with you to accomplish this? I think if we do > this right we can have a really nice phone with lots of options. > > Thanks, > /b > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > >> FreeSWITCH is running nicely on OS X. I have used it since July >> 2006 on my intel Macs with great success. >> I am also developing a GUI application using Cocoa. I started that >> a year ago, but haven't looked at it for a while, but this >> Christmas I have started working on it again. >> >> Ivan >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ivan at myrvold.org Tue Dec 29 11:40:52 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 20:40:52 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> Yes, I would like to participate on this. I have lot of experience with Cocoa on Mac, so I could help with that platform. But I am mostly a GUI programmer, Objective-C my language. But if this is OK with you, I would love to help out here. Ivan Den 29. des. 2009 kl. 18.11 skrev Brian West: > Ivan, > I have been trying to gather up everyone to start a FreeSWITCH based softphone project for Mac, Linux and Windows... you think we could collaborate with you to accomplish this? I think if we do this right we can have a really nice phone with lots of options. > > Thanks, > /b > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > >> FreeSWITCH is running nicely on OS X. I have used it since July 2006 on my intel Macs with great success. >> I am also developing a GUI application using Cocoa. I started that a year ago, but haven't looked at it for a while, but this Christmas I have started working on it again. >> >> Ivan >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lon at kickasspixels.com Tue Dec 29 12:01:26 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 29 Dec 2009 12:01:26 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> Message-ID: <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> I have done a little research into this for my employer. You may want to look at: http://www.qutecom.org/ - I think its QT based. http://code.google.com/p/telephone/ - Its pure Cocoa. I use this for all my testing, it lets me initiate up to 8 calls at a time. On Tue, Dec 29, 2009 at 11:40 AM, Ivan C Myrvold wrote: > Yes, I would like to participate on this. I have lot of experience with > Cocoa on Mac, so I could help with that platform. But I am mostly a GUI > programmer, Objective-C my language. > But if this is OK with you, I would love to help out here. > > Ivan > > Den 29. des. 2009 kl. 18.11 skrev Brian West: > > > Ivan, > > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we could > collaborate with you to accomplish this? I think if we do this right we can > have a really nice phone with lots of options. > > > > Thanks, > > /b > > > > On Dec 29, 2009, at 11:06 AM, Ivan C Myrvold wrote: > > > >> FreeSWITCH is running nicely on OS X. I have used it since July 2006 on > my intel Macs with great success. > >> I am also developing a GUI application using Cocoa. I started that a > year ago, but haven't looked at it for a while, but this Christmas I have > started working on it again. > >> > >> Ivan > >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/fc0e78a9/attachment-0002.html From edpimentl at gmail.com Tue Dec 29 12:08:40 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 15:08:40 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> Message-ID: <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> Add me to the app list .... I use MAC mostly ... Also can you list the new (better, gentler) list of commands to install FreeSwitch on a MAC OSX ... ? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/f0057d1d/attachment-0002.html From edpimentl at gmail.com Tue Dec 29 12:18:38 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 15:18:38 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <0CD2AA5C-743C-4415-A6F3-7A490111724A@myrvold.org> <5d3e0dc60912291201x54fece4exc18e9b6ce37ae28d@mail.gmail.com> Message-ID: <9dc4a1670912291218r581d6231wb020158e0dc1f7c1@mail.gmail.com> Here is my personal "refactored" QuteCom ... *File:* AgileSIP-MAC.zip You have 24 hours to retrieve this file at http://datrshare.datr.ws/download/e567e741e54116f488dc7a24a478e498/edpimentl%40gmail.com/AgileSIP-MAC.zip If you like it... i will create one with a FreeSwitch watermark logo for FS... Best, -E Gpro.ws DatR.ws ---------- Forwarded message ---------- From: Lon Baker Date: Tue, Dec 29, 2009 at 3:01 PM Subject: Re: [Freeswitch-users] MacOSX To: freeswitch-users at lists.freeswitch.org I have done a little research into this for my employer. You may want to look at: http://www.qutecom.org/ - I think its QT based. http://code.google.com/p/telephone/ - Its pure Cocoa. I use this for all my testing, it lets me initiate up to 8 calls at a time. On Tue, Dec 29, 2009 at 11:40 AM, Ivan C Myrvold wrote: > Yes, I would like to participate on this. I have lot of experience with > Cocoa on Mac, so I could help with that platform. But I am mostly a GUI > programmer, Objective-C my language. > But if this is OK with you, I would love to help out here. > > Ivan > > Den 29. des. 2009 kl. 18.11 skrev Brian West: > > > Ivan, > > I have been trying to gather up everyone to start a FreeSWITCH > based softphone project for Mac, Linux and Windows... you think we could > collaborate with you to accomplish this? I think if we do this right we can > have a really nice phone with lots of options. > > > > Thanks, > > /b > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/c2a29435/attachment-0002.html From brian at freeswitch.org Tue Dec 29 12:34:45 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 14:34:45 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> Message-ID: <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> I would love to have a FreeSWITCH based softphone for all three platforms... I just feel a project like that would be kick ass. Must work on 32bit and 64bit of Windows, Mac and Linux ... and not suck like most softphones do. /b On Dec 29, 2009, at 2:08 PM, EdPimentl wrote: > Add me to the app list .... I use MAC mostly ... > > Also can you list the new (better, gentler) list of commands to install FreeSwitch on a MAC OSX ... ? > > Thanks in advance, > -E From jerry.richards at teotech.com Tue Dec 29 12:42:20 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 Dec 2009 12:42:20 -0800 Subject: [Freeswitch-users] Bypass Media True Disables MOH Message-ID: When I uncomment the following tag, internally held calls no longer hear MOH. Is there a way to have the above uncommented and still provide MOH to held calls? Best Regards, Jerry From astmac at stillnewt.org Tue Dec 29 13:08:44 2009 From: astmac at stillnewt.org (Martin Joseph) Date: Tue, 29 Dec 2009 13:08:44 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> Message-ID: <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Sounds cool to me also. There is one softphone for OSX that doesn't suck. It's called JackenIAX. A "universal" softphone sounds cool, but not if it's dumbed down with regards to it's OSX integration. Thanks to all, Marty On Dec 29, 2009, at 12:34 PM, Brian West wrote: > I would love to have a FreeSWITCH based softphone for all three > platforms... I just feel a project like that would be kick ass. > > Must work on 32bit and 64bit of Windows, Mac and Linux ... and not > suck like most softphones do. > > /b > > On Dec 29, 2009, at 2:08 PM, EdPimentl wrote: > >> Add me to the app list .... I use MAC mostly ... >> >> Also can you list the new (better, gentler) list of commands to >> install FreeSwitch on a MAC OSX ... ? >> >> Thanks in advance, >> -E > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Dec 29 13:12:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:12:08 -0600 Subject: [Freeswitch-users] Bypass Media True Disables MOH In-Reply-To: References: Message-ID: <9BF355CF-C633-4BF5-BB8B-642DD81936D1@freeswitch.org> But it doesn't go back to bypass after.... Maybe you can post a bounty for that functionality. /b On Dec 29, 2009, at 2:42 PM, Jerry Richards wrote: > > When I uncomment the following tag, internally held calls no longer hear > MOH. > > > > Is there a way to have the above uncommented and still provide MOH to held > calls? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Dec 29 13:14:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:14:17 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Message-ID: Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. /b On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > There is one softphone for OSX that doesn't suck. It's called > JackenIAX. From ron.freeswitch at mcleodnet.com Tue Dec 29 13:29:58 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Tue, 29 Dec 2009 13:29:58 -0800 Subject: [Freeswitch-users] Custom Events Message-ID: I have two inbound event socket sessions. I send a custom event like this on one socket: sendevent CUSTOM Event-Subclass: wpbx::Bcsm Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER and I receive the following reply: Content-Type: command/reply Reply-Text: +OK On the other event socket, I see this: Content-Length: 615 Content-Type: text/event-plain Event-Subclass: wpbx%3A%3ABcsm Event-Name: COMMAND Core-UUID: 2759f3f4-f4b7-11de-9ff0-11851d44d59f FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-29%2013%3A25%3A49 Event-Date-GMT: Tue,%2029%20Dec%202009%2021%3A25%3A49%20GMT Event-Date-Timestamp: 1262121949103034 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1087 Command: sendevent%20CUSTOM Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER I was expecting to see a CUSTOM event, not COMMAND (or maybe a CUSTOM event in addition to a COMMAND event), like what I see with other custom events such as: Content-Length: 911 Content-Type: text/event-plain Event-Subclass: sofia%3A%3Aregister Event-Name: CUSTOM Core-UUID: 5d56384a-ed29-11de-85c6-27ab474dd533 FreeSWITCH-Hostname: ron-laptop FreeSWITCH-IPv4: 192.168.100.132 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-12-29%2010%3A35%3A36 Event-Date-GMT: Tue,%2029%20Dec%202009%2018%3A35%3A36%20GMT Event-Date-Timestamp: 1262111736464194 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_handle_register Event-Calling-Line-Number: 1127 Event-Subclass: sofia%3A%3Aregister profile-name: internal from-user: 698 from-host: 192.168.100.132 presence-hosts: 192.168.100.132 contact: %22user%22%20%3Csip%3A698%40192.168.100.130%3A5060%3E call-id: 1909944913%40192.168.100.130 rpid: unknown statusd: Registered(UDP) expires: 60 to-user: 698 to-host: 192.168.100.132 network-ip: 192.168.100.130 network-port: 5060 username: 698 realm: 192.168.100.132 user-agent: UTSTARCOM%20F3000/Device%20ID-F3000_TEST Am I doing something wrong, or am I missing something? Thanks, Ron From ivan at myrvold.org Tue Dec 29 13:35:08 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 22:35:08 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> Message-ID: <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> I am using iSoftPhone, works great with FreeSWITCH. Ivan Den 29. des. 2009 kl. 22.14 skrev Brian West: > Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. > > /b > > On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > >> There is one softphone for OSX that doesn't suck. It's called >> JackenIAX. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 13:45:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 15:45:44 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> Message-ID: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? /b On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > I am using iSoftPhone, works great with FreeSWITCH. > > Ivan > > Den 29. des. 2009 kl. 22.14 skrev Brian West: > >> Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >> >> /b >> >> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >> >>> There is one softphone for OSX that doesn't suck. It's called >>> JackenIAX. > From ivan at myrvold.org Tue Dec 29 13:59:31 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 29 Dec 2009 22:59:31 +0100 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Message-ID: <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Yes, I am for this. Happy to join such a project. Ivan Den 29. des. 2009 kl. 22.45 skrev Brian West: > Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? > > /b > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > >> I am using iSoftPhone, works great with FreeSWITCH. >> >> Ivan >> >> Den 29. des. 2009 kl. 22.14 skrev Brian West: >> >>> Does it only do IAX? If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >>> >>> /b >>> >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >>> >>>> There is one softphone for OSX that doesn't suck. It's called >>>> JackenIAX. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 15:06:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 17:06:14 -0600 Subject: [Freeswitch-users] Cisco 501's Message-ID: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> Anyone have access to these phones? Two of them if possible and provisioning information? Thanks, Brian From jerry.richards at teotech.com Tue Dec 29 15:17:56 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 Dec 2009 15:17:56 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail In-Reply-To: <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> Message-ID: I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Monday, December 28, 2009 3:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/0668d901/attachment-0002.html From mike at jerris.com Tue Dec 29 15:37:10 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 29 Dec 2009 18:37:10 -0500 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail In-Reply-To: References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com> <191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> Message-ID: <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> try these drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz Mike On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote: > I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. > > Best Regards, > Jerry > > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Monday, December 28, 2009 3:31 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail > > you have to update the sangoma driver and probably FreeSWITCH for good measure. > Its a known bug in the sangoma driver that has been fixed it the latest release. > > > > On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: > Hello All, > > I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. > > I am still having the problem where a PSTN-to-Internal call via a Sangoma > A101D card stops ringing the internal phone after about 10 seconds. It > should be ringing for 30 seconds and then go to Voice Mail (as an > Internal-to-Internal call does). > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Tuesday, December 22, 2009 8:02 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail > > > I have a Freeswitch PBX server with an installed Sangoma A101D card > connected to a PRI. Most everything works okay, however when I get an > inbound call from the PSTN, if the call is not answered within about 12 > seconds, the call ends (so it doesn't go to voice mail). If I make a call > from one internal phone to another, then it will go to voice mail after 30 > seconds. How can I get the external call to route to voice mail after 30 > seconds? > > I put a new 11595 log into the pastebin. Do you know any Freeswitch setting > that might cause this? > > If this issue has been addressed before, what string should I use to search > for it, because I can't find it. > > Thanks, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/3ac0c607/attachment-0002.html From jmesquita at freeswitch.org Tue Dec 29 16:44:43 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 29 Dec 2009 22:44:43 -0200 Subject: [Freeswitch-users] MacOSX In-Reply-To: <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: Why don't we evolve FSGui to be a softphone? I could use a couple of experienced programmers to help out with it since I pretty much suck at it... FSGui is extensible using plugins so I think that a softphone would be nothing more then just another plugin. Math? Would you like to join in there and put a bit of work with me? Jo?o Mesquita FreeSWITCH? Solutions t: +1 (646) 4959927 On Tue, Dec 29, 2009 at 7:59 PM, Ivan C Myrvold wrote: > Yes, I am for this. Happy to join such a project. > > Ivan > > Den 29. des. 2009 kl. 22.45 skrev Brian West: > > > Guessing the biggest issue is I want to create a softphone project using > FreeSWITCH as the core of the project... is this something people would be > interested in joining? > > > > /b > > > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > > > >> I am using iSoftPhone, works great with FreeSWITCH. > >> > >> Ivan > >> > >> Den 29. des. 2009 kl. 22.14 skrev Brian West: > >> > >>> Does it only do IAX? If so we'll need someone to re-write an IAX2 > stack since the libiax2 from Digium is no longer updated to keep pace with > Asterisk and is now incompatible. Which is the main reason we are thinking > about dropping IAX support unless someone writes a license compatible lib or > updates and takes over mod_iax aka owns it as their own. > >>> > >>> /b > >>> > >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: > >>> > >>>> There is one softphone for OSX that doesn't suck. It's called > >>>> JackenIAX. > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/8b42de37/attachment-0002.html From brian at freeswitch.org Tue Dec 29 16:51:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 18:51:44 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: Sounds like a plan to me... who wants to take the lead on the project... we'll host it.. setup SVN, provide jira access, fisheye and wiki space... /b On Dec 29, 2009, at 6:44 PM, Jo?o Mesquita wrote: > Why don't we evolve FSGui to be a softphone? I could use a couple of experienced programmers to help out with it since I pretty much suck at it... FSGui is extensible using plugins so I think that a softphone would be nothing more then just another plugin. > > Math? Would you like to join in there and put a bit of work with me? > > Jo?o Mesquita > FreeSWITCH? Solutions > t: +1 (646) 4959927 From edpimentl at gmail.com Tue Dec 29 17:29:51 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 29 Dec 2009 20:29:51 -0500 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> Message-ID: <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> For starters, one can use use the attached pics as the springboard for the future design of the softphone. -E -------------- next part -------------- An HTML attachment was scrubbed... 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Name: Picture 3.png Type: image/png Size: 32037 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/2e837688/attachment-0005.png From dujinfang at gmail.com Tue Dec 29 17:36:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 30 Dec 2009 09:36:20 +0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> Message-ID: <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> I had wrote a Air based GUI, is it make sense? http://wiki.freeswitch.org/wiki/FsAir 2009/12/30 Brian West : > Guessing the biggest issue is I want to create a softphone project using FreeSWITCH as the core of the project... is this something people would be interested in joining? > > /b > > On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote: > >> I am using iSoftPhone, works great with FreeSWITCH. >> >> Ivan >> >> Den 29. des. 2009 kl. 22.14 skrev Brian West: >> >>> Does it only do IAX? ?If so we'll need someone to re-write an IAX2 stack since the libiax2 from Digium is no longer updated to keep pace with Asterisk and is now incompatible. ?Which is the main reason we are thinking about dropping IAX support unless someone writes a license compatible lib or updates and takes over mod_iax aka owns it as their own. >>> >>> /b >>> >>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote: >>> >>>> There is one softphone for OSX that doesn't suck. ?It's called >>>> JackenIAX. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Dec 29 17:50:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 29 Dec 2009 19:50:07 -0600 Subject: [Freeswitch-users] MacOSX In-Reply-To: <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> Message-ID: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> You should join IRC and join in JM and really start the official softphone project. /b On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > I had wrote a Air based GUI, is it make sense? > > http://wiki.freeswitch.org/wiki/FsAir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/39c4ce35/attachment-0002.html From lon at kickasspixels.com Tue Dec 29 18:30:28 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 29 Dec 2009 18:30:28 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> Message-ID: I would be happy to provide project managent, user testing and ui design to this effort. Lon Sent from my iPhone On Dec 29, 2009, at 5:50 PM, Brian West wrote: > You should join IRC and join in JM and really start the official > softphone project. > > /b > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > >> I had wrote a Air based GUI, is it make sense? >> >> http://wiki.freeswitch.org/wiki/FsAir > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091229/e0d04a6f/attachment-0002.html From ron.freeswitch at mcleodnet.com Tue Dec 29 21:23:33 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Tue, 29 Dec 2009 21:23:33 -0800 Subject: [Freeswitch-users] PHP ESL Problem Message-ID: <285BD733E19541989B31B95871BF5642@fromage> Would someone please take a look at this simple PHP event socket script and tell me what I am doing wrong - or tell me that this could be a bug elsewhere? Any help would be appreciated. When I run the script without the call to execute(), everything seems fine. When I include the call to execute(), the calls to getType() return CUSTOM for a while, then later start to return the correct name. #!/usr/bin/php events('plain', 'ALL'); // call endpoint, get uuid $event = $eventSocket->api('originate', $endPoint . ' &park'); $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($dummy, $uuid) = explode('+OK ', $eventLine); if ($uuid) { break; } } // play announcement to endpoint $event = $eventSocket->execute('playback', '/opt/ann/user-busy.wav', $uuid); // monitor events while (TRUE) { echo "getType: " . $event->getType() . "\n"; $serializedEvent = explode("\n", $event->serialize()); foreach ($serializedEvent as $eventLine) { list($header, $value) = explode(': ', $eventLine); if ($header == "Event-Name") { printf($eventLine . "\n"); } if ($header == "Content-Type") { printf($eventLine . "\n"); } } printf("\n"); $event = $eventSocket->recvEvent(); }?> Run without the call to execute(): ================================== getType: CUSTOM Content-Type: api/response getType: CHANNEL_CREATE Event-Name: CHANNEL_CREATE getType: CHANNEL_OUTGOING Event-Name: CHANNEL_OUTGOING getType: CHANNEL_ORIGINATE Event-Name: CHANNEL_ORIGINATE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CHANNEL_PROGRESS Event-Name: CHANNEL_PROGRESS getType: HEARTBEAT Event-Name: HEARTBEAT getType: HEARTBEAT Event-Name: RE_SCHEDULE getType: CALL_UPDATE Event-Name: CALL_UPDATE getType: CODEC Event-Name: CODEC getType: CODEC Event-Name: CODEC getType: CHANNEL_ANSWER Event-Name: CHANNEL_ANSWER getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: API Event-Name: API getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_PARK Event-Name: CHANNEL_PARK getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Run with the call to execute(): =============================== getType: CUSTOM Content-Type: command/reply getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_CREATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_OUTGOING getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ORIGINATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PROGRESS getType: CUSTOM Content-Type: text/event-plain Event-Name: CALL_UPDATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CODEC getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_ANSWER getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: API getType: CUSTOM Content-Type: text/event-plain Event-Name: PRESENCE_IN getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_STATE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_EXECUTE getType: CUSTOM Content-Type: text/event-plain Event-Name: CHANNEL_PARK getType: CHANNEL_EXECUTE Event-Name: CHANNEL_EXECUTE getType: CHANNEL_HANGUP Event-Name: CHANNEL_HANGUP getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: COMMAND Event-Name: COMMAND getType: CHANNEL_UNPARK Event-Name: CHANNEL_UNPARK getType: CHANNEL_EXECUTE_COMPLETE Event-Name: CHANNEL_EXECUTE_COMPLETE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: PRESENCE_IN Event-Name: PRESENCE_IN getType: CHANNEL_HANGUP_COMPLETE Event-Name: CHANNEL_HANGUP_COMPLETE getType: CHANNEL_STATE Event-Name: CHANNEL_STATE getType: CHANNEL_DESTROY Event-Name: CHANNEL_DESTROY getType: CHANNEL_STATE Event-Name: CHANNEL_STATE Thanks, Ron From sharad at coraltele.com Tue Dec 29 22:17:21 2009 From: sharad at coraltele.com (Sharad) Date: Tue, 29 Dec 2009 22:17:21 -0800 (PST) Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions Message-ID: <1262153841443-4231122.post@n2.nabble.com> Hi I just want to know what should be the approx configuration of the server for 50 concurrent call sessions having 3000-4000 users. Regards -- View this message in context: http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231122.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 29 22:36:04 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 00:36:04 -0600 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262153841443-4231122.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> Message-ID: <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> Amplify Query... not enough data to make a logical compilation of requested data. /b On Dec 30, 2009, at 12:17 AM, Sharad wrote: > Hi > > I just want to know what should be the approx configuration of the server > for 50 concurrent call sessions having 3000-4000 users. > > Regards From sharad at coraltele.com Tue Dec 29 22:55:19 2009 From: sharad at coraltele.com (Sharad) Date: Tue, 29 Dec 2009 22:55:19 -0800 (PST) Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> Message-ID: <1262156119067-4231210.post@n2.nabble.com> Thanks /b for your kind reply... I want to integrate the freeswitch only for Auto Attendant & voicemail with a SIP server. All the users (approx 5000) shall be registered with SIP server. Whenever auto attendant or voicemail will be required, SIP server will establish the SIP trunking with Freeswitch. So I am assuming that freeswitch may get 50-60 calls at a time either for auto attendant or voicemail. Whenever there is a call with freeswitch for auto attendant, caller will punch the desired user no. & than freeswitch will throw the call back to SIP server. in this case, RTP will not be flowing through freeswitch server. So just want to know what should be the PC hardware specifications for this call traffic. I am willing to buy a good server hardware for this. So plz advice... Regards Sharad Brian West wrote: > > Amplify Query... not enough data to make a logical compilation of > requested data. > > /b > > On Dec 30, 2009, at 12:17 AM, Sharad wrote: >> Hi >> >> I just want to know what should be the approx configuration of the server >> for 50 concurrent call sessions having 3000-4000 users. >> >> Regards > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231210.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Tue Dec 29 23:19:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 01:19:53 -0600 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262156119067-4231210.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> Message-ID: <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> Might I recommend a project that was written for that exact purpose? Have you heard of OpenSIPS/OpenSER or what is that other name... I can't spell it nor remember it... but it combined with FreeSWITCH would solve your problem better I suspect. Use the proxy for registrations and FreeSWITCH for media servers. /b On Dec 30, 2009, at 12:55 AM, Sharad wrote: > > Thanks /b for your kind reply... > > I want to integrate the freeswitch only for Auto Attendant & voicemail with > a SIP server. All the users (approx 5000) shall be registered with SIP > server. Whenever auto attendant or voicemail will be required, SIP server > will establish the SIP trunking with Freeswitch. > > So I am assuming that freeswitch may get 50-60 calls at a time either for > auto attendant or voicemail. Whenever there is a call with freeswitch for > auto attendant, caller will punch the desired user no. & than freeswitch > will throw the call back to SIP server. in this case, RTP will not be > flowing through freeswitch server. > > So just want to know what should be the PC hardware specifications for this > call traffic. > > I am willing to buy a good server hardware for this. So plz advice... > > Regards > Sharad From talk2ram at gmail.com Tue Dec 29 23:20:38 2009 From: talk2ram at gmail.com (ram) Date: Wed, 30 Dec 2009 12:50:38 +0530 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <1262156119067-4231210.post@n2.nabble.com> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> Message-ID: how about this URL http://wiki.freeswitch.org/wiki/Specsheet#Minimum.2FRecommended_System_Requirements Ram On Wed, Dec 30, 2009 at 12:25 PM, Sharad wrote: > > Thanks /b for your kind reply... > > I want to integrate the freeswitch only for Auto Attendant & voicemail with > a SIP server. All the users (approx 5000) shall be registered with SIP > server. Whenever auto attendant or voicemail will be required, SIP server > will establish the SIP trunking with Freeswitch. > > So I am assuming that freeswitch may get 50-60 calls at a time either for > auto attendant or voicemail. Whenever there is a call with freeswitch for > auto attendant, caller will punch the desired user no. & than freeswitch > will throw the call back to SIP server. in this case, RTP will not be > flowing through freeswitch server. > > So just want to know what should be the PC hardware specifications for this > call traffic. > > I am willing to buy a good server hardware for this. So plz advice... > > Regards > Sharad > > > > Brian West wrote: > > > > Amplify Query... not enough data to make a logical compilation of > > requested data. > > > > /b > > > > On Dec 30, 2009, at 12:17 AM, Sharad wrote: > >> Hi > >> > >> I just want to know what should be the approx configuration of the > server > >> for 50 concurrent call sessions having 3000-4000 users. > >> > >> Regards > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/Server-Configuration-for-50-concurrent-sessions-tp4231122p4231210.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/b3318f36/attachment-0002.html From talk2ram at gmail.com Wed Dec 30 00:19:57 2009 From: talk2ram at gmail.com (ram) Date: Wed, 30 Dec 2009 13:49:57 +0530 Subject: [Freeswitch-users] Server Configuration for 50 concurrent sessions In-Reply-To: <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> References: <1262153841443-4231122.post@n2.nabble.com> <7DB6936B-D5A5-4E61-93C4-D74309A5A040@freeswitch.org> <1262156119067-4231210.post@n2.nabble.com> <9C69A12E-29B0-40AA-A5EA-A858A5F4D44D@freeswitch.org> Message-ID: On Wed, Dec 30, 2009 at 12:49 PM, Brian West wrote: > Might I recommend a project that was written for that exact purpose? Have > you heard of OpenSIPS/OpenSER or what is that other name... I can't spell it > nor remember it... but it combined with FreeSWITCH would solve your problem > better I suspect. Use the proxy for registrations and FreeSWITCH for media > servers. > > /b > yes you can use Opensips to Loadbalance N (X) Freeswitch Boxes for loadbalance to meet high volume calls Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a4abc2a6/attachment-0002.html From msc at freeswitch.org Wed Dec 30 00:43:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 00:43:16 -0800 Subject: [Freeswitch-users] Cisco 501's In-Reply-To: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> References: <21A22B12-5FEB-4359-90D4-6795651CA80B@freeswitch.org> Message-ID: <87f2f3b90912300043t111582b9y602aeb88f4ea9126@mail.gmail.com> I have one in my garage collecting dust. :) On Tue, Dec 29, 2009 at 3:06 PM, Brian West wrote: > Anyone have access to these phones? Two of them if possible and > provisioning information? > > Thanks, > Brian > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/7c8d0b8e/attachment-0002.html From mcampbellsmith at gmail.com Wed Dec 30 01:42:09 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Wed, 30 Dec 2009 20:42:09 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> Message-ID: <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> Hi Brian, But that is all I see... I don't see any further messages after the 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that this is part of the authentication procedure, but it stops at the 401/Authorized message What about in the pastebin? Do you see anything there? This is when the call gets setup, but the IP address in the SDP says a private IP address http://pastebin.freeswitch.org/11632 On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: > If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. ?I would have to see what happens after the 401 to see if it really did fail. > > /b > > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > >> This is all I see and then registration fails. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From codecomplete at free.fr Wed Dec 30 03:23:17 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 03:23:17 -0800 (PST) Subject: [Freeswitch-users] Port item in "sofia status" output? Message-ID: <26965939.post@talk.nabble.com> Hello I was wondering: What is the meaning of the Port item that shows up when I run "sofia status"? > sofia status profile internal reg [...] IP: 192.168.0.1 Port: 59724 Is is the UDP port on which the phone listens for incoming SIP messages, or is it the RTP port on which it expects to receive voice packets from the remote phone to which is it will be connected? BTW, does RTP use a single port to TX/RX voice, or does it use the RTP port and the one that immediately follows? Thank you. -- View this message in context: http://old.nabble.com/Port-item-in-%22sofia-status%22-output--tp26965939p26965939.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From alhakeem at gmail.com Wed Dec 30 03:38:10 2009 From: alhakeem at gmail.com (Abdul Hakeem) Date: Wed, 30 Dec 2009 11:38:10 -0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <6E6726FE-4CB1-4E76-8542-068F2330F6F9@myrvold.org> <9dc4a1670912291729k4a475275mc28a45d157057d63@mail.gmail.com> Message-ID: <001801ca8944$91036260$b30a2720$@com> Is anyone working on this or similar soft-phone project ? Please share with us ? Cheers, AH From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of EdPimentl Sent: Wednesday, December 30, 2009 1:30 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] MacOSX For starters, one can use use the attached pics as the springboard for the future design of the softphone. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/d1ccc908/attachment-0002.html From codecomplete at free.fr Wed Dec 30 04:03:06 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:03:06 -0800 (PST) Subject: [Freeswitch-users] [Dialplan?] Call either sent to VM or phone doesn't ring at all In-Reply-To: <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> References: <26892767.post@talk.nabble.com> <26903707.post@talk.nabble.com> <87f2f3b90912231110k767b1d10r443946930aac5155@mail.gmail.com> Message-ID: <26966693.post@talk.nabble.com> Thanks for the tip. I ended up re-installing CentOS followed by compiling the latest SVN source, and I'm back to normal. The only possible cause I see is that changing the PSU sometimes requires flushing the CMOS, but as to why this would cause Freeswitch to route calls to VM, I have no idea. Sorry for the confusion. -- View this message in context: http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26966693.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 30 04:07:38 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:07:38 -0800 (PST) Subject: [Freeswitch-users] freeswitch init In-Reply-To: References: Message-ID: <26966731.post@talk.nabble.com> I'll take advantage of this thread to ask whether it's OK/recommended to create a new user/group "freeswitch", and "chown -Rf /usr/local/freeswitch"? Thank you. -- View this message in context: http://old.nabble.com/freeswitch-init-tp26926152p26966731.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Wed Dec 30 04:18:35 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 04:18:35 -0800 (PST) Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch In-Reply-To: <20091229052106.2592813F5@sinclaire.sibble.net> References: <20091229052106.2592813F5@sinclaire.sibble.net> Message-ID: <26966821.post@talk.nabble.com> Harondel J. Sibble wrote: > I am trying to get 2x X100P's working Provided you are aware that a lot of users have problems getting X100xP cards to work on their hardware (either due to some issue with the PCI bus, or the Silicon Labs DAA chips not supporting the POTS line to which they are connected)... here's what I did to compile Dahdi (the new name for Zaptel) from source: 1. Download and unpack the Dahdi tarball 2. make all ; make install ; make config 3. cd /etc/dahdi/ ; vi system.conf: #Per FS wiki: "Tones should be configured in OpenZAP and not in zaptel. FreeSWITCH uses its libteletone for tones generation and detection, and does not rely on zaptel tones configuration. Therefore it does not matter which country zone is configured in zaptel. Make sure that you have loaded your country-specific tones at /etc/openzap/tones.conf" #loadzone = fr #defaultzone = fr fxsks=1 4. vi modules; add "wctdm" 5. /etc/init.d/dahdi start 6. lsmod to check loaded modules 7. dahdi_cfg -vvv ; dahdi_scan ; dahdi_test -v 8. ls -la /proc/dahdi/ ; cat /proc/dahdi/1 The next step, which I haven't done yet, is compiling OpenZap so it talks to Dahdi/Zaptel, which talks to the actual hardware. -- View this message in context: http://old.nabble.com/problems-getting-openzap-compiled-for-use-with-freeswitch-tp26951344p26966821.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From petedao at gmail.com Wed Dec 30 05:14:03 2009 From: petedao at gmail.com (Pete Kay) Date: Wed, 30 Dec 2009 21:14:03 +0800 Subject: [Freeswitch-users] freeswitch and H323 Message-ID: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> Hi, has anyone been able to get H323 to work? I have problem trying to get it compiled with either 1.0.4 or 1.0.5. Thanks, pete From Russell.Mosemann at cune.org Wed Dec 30 05:57:43 2009 From: Russell.Mosemann at cune.org (Russell Mosemann) Date: Wed, 30 Dec 2009 07:57:43 -0600 Subject: [Freeswitch-users] freeswitch init In-Reply-To: <26966731.post@talk.nabble.com> References: <26966731.post@talk.nabble.com> Message-ID: <953218C0A63749C4BF083829F5372E5E@cune.pri> Fred-145 wrote: > I'll take advantage of this thread to ask whether it's OK/recommended to > create a new user/group "freeswitch", and "chown -Rf > /usr/local/freeswitch"? Yes. -- Russell Mosemann From anthony.minessale at gmail.com Wed Dec 30 06:04:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 08:04:39 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> Message-ID: <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> are you using more than one profile here? if so can you repeat the trace with siptrace on in both profiles. I notice this on the trace: 1. Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 2. From: 1000 ;tag=70c4b62bd9443c9fo0 and this: sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000 at 192.168.1.120] 192.168.1 .120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20 Are you behind double nat and or is FS also behind nat? To address the gentleman who mentioned he was spoiled by asterisk. Keep in mind we have several advanced nat techniques but many of them break other situations and they are not all turned on at once with one parameter like in asterisk, we prefer you use the appropriate ones based on the situation. On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi Brian, > > But that is all I see... I don't see any further messages after the > 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that > this is part of the authentication procedure, but it stops at the > 401/Authorized message > > What about in the pastebin? Do you see anything there? This is when > the call gets setup, but the IP address in the SDP says a private IP > address http://pastebin.freeswitch.org/11632 > > On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: > > If you're using the 401 as an indication that it fails then you don't > understand how digest authentication works. I would have to see what > happens after the 401 to see if it really did fail. > > > > /b > > > > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: > > > >> This is all I see and then registration fails. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/3cd5a327/attachment-0002.html From dujinfang at gmail.com Wed Dec 30 06:17:39 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 30 Dec 2009 22:17:39 +0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <52FDBF06-4167-4B4B-9587-61531B0EEA94@freeswitch.org> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> Message-ID: <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> I rarely joined in IRC, becuase I live in China, timezone +8000 .... I really would like to start the official softphone, btw, what is JM? 2009/12/30, Brian West : > You should join IRC and join in JM and really start the official softphone > project. > > /b > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > >> I had wrote a Air based GUI, is it make sense? >> >> http://wiki.freeswitch.org/wiki/FsAir > > From jmesquita at freeswitch.org Wed Dec 30 06:27:40 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 30 Dec 2009 12:27:40 -0200 Subject: [Freeswitch-users] MacOSX In-Reply-To: <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <9dc4a1670912291208j937ced8n551eefd97fa179af@mail.gmail.com> <2D505043-E1CD-4BAD-B829-F983CC7522FD@freeswitch.org> <2B898305-194A-4E9F-971E-F6FCC1B1FF33@stillnewt.org> <0351BD70-B846-49B4-ADE8-9AA910B929C0@myrvold.org> <0F413883-8D86-43A7-A7A2-01B6C9A8CCBB@freeswitch.org> <23f91030912291736l40570c1at5f1005711b52252c@mail.gmail.com> <60B8C5F1-59F4-4621-82F1-2FEAE5A4993E@freeswitch.org> <23f91030912300617gfbfd372l9903f942dd789e14@mail.gmail.com> Message-ID: What is JM is not the question but rather WHO is JM and that would be me. :-) I have already stripped down the config handler based on mod_xml_curl. I have been discussing with Brian how to make it happen and I am conducting a couple of tests with Qt. Today I might be able to have it properly linked with Qt and the core spawn on its own thread inside the Qt event loop. I'll keep you posted. Jo?o Mesquita A.K.A -> JM On Wed, Dec 30, 2009 at 12:17 PM, Seven Du wrote: > I rarely joined in IRC, becuase I live in China, timezone +8000 .... > I really would like to start the official softphone, btw, what is JM? > > 2009/12/30, Brian West : > > You should join IRC and join in JM and really start the official > softphone > > project. > > > > /b > > > > On Dec 29, 2009, at 7:36 PM, Seven Du wrote: > > > >> I had wrote a Air based GUI, is it make sense? > >> > >> http://wiki.freeswitch.org/wiki/FsAir > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cad80cc2/attachment-0002.html From lei.tlfly at gmail.com Wed Dec 30 06:38:44 2009 From: lei.tlfly at gmail.com (Lei Tang) Date: Wed, 30 Dec 2009 22:38:44 +0800 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> Message-ID: <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> Hi Anthony and Brian, I have grep some sip trace log, Could you please take a look at the attachments? lega.pcap is the log of a leg, legb.pcap is the log of b leg. It seems fs never sent the reinvite request to b, and sent 200ok response without sdp content to a immediately. My scenario is as follow, and I'm using proxy media mode. invite invite a <------------> FS<---------------> b 2009/12/30 Anthony Minessale > We now disable sofia SOA mode during proxy calls. > This means that sofia will not try to get involved in the media negotiation > at all which is the optimal behavior. > Previous versions would butt in and try to fix the error but now it just > stays out of the way. > > You can see in your trace that the device sends a packet with no SDP > therefore so does sofia. > > You can either turn off proxy-media or post a bounty for me to go hack a > workaround into the patch I spent many hours on getting things to work > right. Whatever you experienced with 1.0.4 was a happy coincidence where > sofia was fixing a bug in your phone for you. > > > > > On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: > >> This means there was no sdp sent. Did you confirm this with siptrace? >> >> On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: >> >> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, following >> code in sofia.c send the 200ok response >> sofia.c >> function sofia_handle_sip_i_state .... >> ......... >> switch(ss_state) >> ................ >> case nua_callstate_received: >> ..................... >> else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && >> !r_sdp) { >> nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); >> sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); >> goto done; >> } >> >> The cause is r_sdp is null, but I don't known why tl_gets don't return >> remote sdp tag, it's quite strange. >> >> 2009/12/29 Brian West < brian at freeswitch.org> >> >>> the 200ok is not from FS.. its from the end point... so its not us thats >>> not putting the SDP into the 200ok but the device you're talking to because >>> in proxy media they are passed as is. >>> >>> /b >>> >>> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >>> >>> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >>> sip agent I'm using is x-lite and wxCommunicator. >>> > I will test if trunk 16055 work when I set proxy media mode to false >>> tomorrow. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Lei.Tang >> lei.tlfly at gmail.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Lei.Tang lei.tlfly at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: lega.pcap Type: application/octet-stream Size: 15372 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0004.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: legb.pcap Type: application/octet-stream Size: 9395 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/a52a2ace/attachment-0005.obj From anthony.minessale at gmail.com Wed Dec 30 07:19:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 09:19:26 -0600 Subject: [Freeswitch-users] Hold is broken in trunk 16055 In-Reply-To: <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> References: <50c41b4e0912282314k17e29414x15736401ae40bd59@mail.gmail.com> <50c41b4e0912290653q1a8be612kfe274963212da109@mail.gmail.com> <954AED85-8DCE-46A3-B803-07564C6CC95F@freeswitch.org> <50c41b4e0912290737u57611936g1338a7ba5e6f13ee@mail.gmail.com> <155946E2-7DB6-43D1-9418-BB571F962ABF@jerris.com> <191c3a030912290845pe601ceby5e1a9945031b9869@mail.gmail.com> <50c41b4e0912300638p62a9dc7cn28812f430ca667a8@mail.gmail.com> Message-ID: <191c3a030912300719m5557ad3cyad1f483643001f67@mail.gmail.com> I was wondering if you read my last email? your phone is not sending an SDP, but you expect FS to magically pass it on with an SDP? Did you try anything I said? I feel that you have ignored my 2 paragraph explanation. On Wed, Dec 30, 2009 at 8:38 AM, Lei Tang wrote: > Hi Anthony and Brian, I have grep some sip trace log, Could you please take > a look at the attachments? lega.pcap is the log of a leg, legb.pcap is the > log of b leg. It seems fs never sent the reinvite request to b, and sent > 200ok response without sdp content to a immediately. > My scenario is as follow, and I'm using proxy media mode. > > invite invite > a <------------> FS<---------------> b > > 2009/12/30 Anthony Minessale > > We now disable sofia SOA mode during proxy calls. >> This means that sofia will not try to get involved in the media >> negotiation at all which is the optimal behavior. >> Previous versions would butt in and try to fix the error but now it just >> stays out of the way. >> >> You can see in your trace that the device sends a packet with no SDP >> therefore so does sofia. >> >> You can either turn off proxy-media or post a bounty for me to go hack a >> workaround into the patch I spent many hours on getting things to work >> right. Whatever you experienced with 1.0.4 was a happy coincidence where >> sofia was fixing a bug in your phone for you. >> >> >> >> >> On Tue, Dec 29, 2009 at 10:08 AM, Michael Jerris wrote: >> >>> This means there was no sdp sent. Did you confirm this with siptrace? >>> >>> On Dec 29, 2009, at 10:37 AM, Lei Tang wrote: >>> >>> Hi Brian, I don't think so, I have debuged fs, If I'm not wrong, >>> following code in sofia.c send the 200ok response >>> sofia.c >>> function sofia_handle_sip_i_state .... >>> ......... >>> switch(ss_state) >>> ................ >>> case nua_callstate_received: >>> ..................... >>> else if (tech_pvt && sofia_test_flag(tech_pvt, TFLAG_SDP) && >>> !r_sdp) { >>> nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); >>> sofia_set_flag_locked(tech_pvt, TFLAG_NOSDP_REINVITE); >>> goto done; >>> } >>> >>> The cause is r_sdp is null, but I don't known why tl_gets don't return >>> remote sdp tag, it's quite strange. >>> >>> 2009/12/29 Brian West < brian at freeswitch.org> >>> >>>> the 200ok is not from FS.. its from the end point... so its not us thats >>>> not putting the SDP into the 200ok but the device you're talking to because >>>> in proxy media they are passed as is. >>>> >>>> /b >>>> >>>> On Dec 29, 2009, at 8:53 AM, Lei Tang wrote: >>>> >>>> > Hi Brian, thanks for your help, I am using FS in proxy media mode. the >>>> sip agent I'm using is x-lite and wxCommunicator. >>>> > I will test if trunk 16055 work when I set proxy media mode to false >>>> tomorrow. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Lei.Tang >>> lei.tlfly at gmail.com >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Lei.Tang > lei.tlfly at gmail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/5c16463a/attachment-0002.html From codecomplete at free.fr Wed Dec 30 07:29:12 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 30 Dec 2009 07:29:12 -0800 (PST) Subject: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch In-Reply-To: <4B3142F7.1080600@skypro.be> References: <4B3142F7.1080600@skypro.be> Message-ID: <26968615.post@talk.nabble.com> Kristoff Bonne-2 wrote: > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" > device for just 15 euro. This is a device which has on one side a > USB-connector and on the other side 2 RJ-11 connectors (one FXO and one > FSX). Internally, the device seams to contain a tigerjet 560C chipset. > (see here: http://www.tjnet.com/chips/tiger560C.htm) It looks like one of those ATA boxes that has an FXO port so you can make calls either through a VoIP provider or through your landline. Now, if it can also act as an SIP/PSTN gateway, I'd be very interested in that ?15 piece of hardware ;-) -- View this message in context: http://old.nabble.com/tigerjet-560C-USB-to-rj11%3A-incorperate-usbhid-usbsnd-device-into-freeswitch-tp26895402p26968615.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From fdenkens at ilibris.be Wed Dec 30 08:15:33 2009 From: fdenkens at ilibris.be (Frederik Denkens | iLibris) Date: Wed, 30 Dec 2009 17:15:33 +0100 Subject: [Freeswitch-users] Freeswitch 1.0.4 and Patton 4554 gateway registration problem Message-ID: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> Hi all! We our looking forward to replacing our legacy PBX with a Freeswitch platform, but are struggling with getting our Patton SmartNode 4554 (SIP/ISDN) gateway hooked up. Any assistance from the community would be great! First step is getting Freeswitch and the Patton talking to each other. Goal? To be able to have incoming and outgoing calls going over the Patton to the ISDN network. Internal PBX: - Freeswitch 1.0.4 installed from tar with default (demo) setup on IP 10.156.10.93 - Patton 4554 with simple config on IP 10.156.10.90 We set it up that the Patton registers with the PBX and we get the error: --------------------- 2009-12-30 17:07:08.11907 [WARNING] sofia_reg.c:1771 Can't find user [101 at 10.156.10.93 ] You must define a domain called '10.156.10.93' in your directory and add a user with the id="101" attribute and you must configure your device to use the proper domain in it's authentication credentials. --------------------- Find more info below. So any help would be great! Many thanks! Frederik Denkens Belgium +32 475 96 04 93 We defined a gateway in conf/sip_profiles/external/patton.xml: --------------------- --> --------------------- And the relevant parts of the Patton config: --------------------- # define auth authentication-service AUTH_SVC username 101 password 101 # patton registers location-service LOCATION_SVC domain 1 10.156.10.93 identity 101 authentication outbound authenticate 1 authentication-service AUTH_SVC username 101 registration outbound registrar 10.156.10.93 5080 lifetime 3600 register auto context sip-gateway GW_SIP interface IF_SIP bind interface IF_IP_WAN context router port 5060 context sip-gateway GW_SIP bind location-service LOCATION_SVC no shutdown --------------------- Output from 'sofia status' --------------------- 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== internal profile sip:mod_sofia at 10.156.10.93:5060 RUNNING (0) external profile sip:mod_sofia at 10.156.10.93:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG 10.156.10.93 alias internal ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== --------------------- sip trace --------------------- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.156.10.90:5060;branch=z9hG4bKe231fe7b97aa2c0cf From: ;tag=b793adfadf To: ;tag=Ua16yjtFgDQ5c Call-ID: add8fbd86264310b CSeq: 12517 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 Before printing this e-mail, please consider the impact on the environment. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/7be51e9e/attachment-0002.html From voippa at gmail.com Wed Dec 30 06:19:54 2009 From: voippa at gmail.com (David Schwartz) Date: Wed, 30 Dec 2009 16:19:54 +0200 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? Message-ID: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> What is the current communication protocol with the media proxy? I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? Thanks, Adino -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/ead6076d/attachment-0002.html From voippa at gmail.com Wed Dec 30 08:09:07 2009 From: voippa at gmail.com (Adino) Date: Wed, 30 Dec 2009 18:09:07 +0200 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? Message-ID: <6e40a4420912300809t26d1d316had55361de65d2f27@mail.gmail.com> What is the current communication protocol with the media proxy? I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/87546a48/attachment-0002.html From brian at freeswitch.org Wed Dec 30 08:23:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 10:23:15 -0600 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? In-Reply-To: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> References: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> Message-ID: <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> Someone could pay to have support added but as it stands we don't have it yet. If its going to be done it has to be proper mgcp support not half ass support. /b On Dec 30, 2009, at 8:19 AM, David Schwartz wrote: > I noticed that MGCP is not on roadmap - can anyone provide a little more info on this please? From help at pdscc.com Wed Dec 30 08:28:48 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Wed, 30 Dec 2009 08:28:48 -0800 Subject: [Freeswitch-users] problems getting openzap compiled for use with freeswitch In-Reply-To: <26966821.post@talk.nabble.com> References: <20091229052106.2592813F5@sinclaire.sibble.net>, <26966821.post@talk.nabble.com> Message-ID: <20091230162847.DAC71174C@sinclaire.sibble.net> On 30 Dec 2009 at 4:18, Fred-145 wrote: > Provided you are aware that a lot of users have problems getting > X100xP cards to work on their hardware (either due to some issue with > the PCI bus, or the Silicon Labs DAA chips not supporting the POTS line > to which they are connected)... here's what I did to compile Dahdi (the > new name for Zaptel) from source: Not an issue in this machine, the 2 cards worked fine with asterisk now and dahdi previously. > The next step, which I haven't done yet, is compiling OpenZap so it > talks to Dahdi/Zaptel, which talks to the actual hardware. I'd love to hear your feedback on that when you do. I'm hoping to get this up and running this weekend, but currently more committed to getting zrtp working with the tivi client on my windows mobile phone. It's giving a registration forbidden message, but I don't see anything on the fs_cli to indicate it's even connected :-( -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From msc at freeswitch.org Wed Dec 30 09:00:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 09:00:18 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly! Message-ID: <87f2f3b90912300900i678303e3ofcbba57fce2a0faa@mail.gmail.com> Please join us today for a special Wednesday edition of the weekly FreeSWITCH conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30 -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cd533d43/attachment-0002.html From ron.freeswitch at mcleodnet.com Wed Dec 30 09:33:34 2009 From: ron.freeswitch at mcleodnet.com (Ron McLeod) Date: Wed, 30 Dec 2009 09:33:34 -0800 Subject: [Freeswitch-users] Custom Events In-Reply-To: References: Message-ID: <754D535B4FC64203B1F8D1670F360F3E@fromage> I found that this works as a work-around: sendevent Event-Name: CUSTOM Event-Subclass: wpbx::Bcsm Bcsm-Operation: Bcsm-Event Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 Bcsm-Event: ANSWER > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod > Sent: Tuesday, December 29, 2009 1:30 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Custom Events > > I have two inbound event socket sessions. > > I send a custom event like this on one socket: > > sendevent CUSTOM > Event-Subclass: wpbx::Bcsm > Bcsm-Operation: Bcsm-Event > Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d > Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 > Bcsm-Event: ANSWER > > and I receive the following reply: > > Content-Type: command/reply > Reply-Text: +OK > > > On the other event socket, I see this: > > Content-Length: 615 > Content-Type: text/event-plain > > Event-Subclass: wpbx%3A%3ABcsm > Event-Name: COMMAND > Core-UUID: 2759f3f4-f4b7-11de-9ff0-11851d44d59f > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-29%2013%3A25%3A49 > Event-Date-GMT: Tue,%2029%20Dec%202009%2021%3A25%3A49%20GMT > Event-Date-Timestamp: 1262121949103034 > Event-Calling-File: mod_event_socket.c > Event-Calling-Function: read_packet > Event-Calling-Line-Number: 1087 > Command: sendevent%20CUSTOM > Bcsm-Operation: Bcsm-Event > Bcsm-Uuid: fd96fe6f-ad88-4882-8691-4f849074554d > Bcsm-Peer-Uuid: edf0604c-f4a4-11de-85c6-27ab474dd533 > Bcsm-Event: ANSWER > > > I was expecting to see a CUSTOM event, not COMMAND (or maybe a CUSTOM > event > in addition to a COMMAND event), like what I see with other custom events > such as: > > Content-Length: 911 > Content-Type: text/event-plain > > Event-Subclass: sofia%3A%3Aregister > Event-Name: CUSTOM > Core-UUID: 5d56384a-ed29-11de-85c6-27ab474dd533 > FreeSWITCH-Hostname: ron-laptop > FreeSWITCH-IPv4: 192.168.100.132 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-12-29%2010%3A35%3A36 > Event-Date-GMT: Tue,%2029%20Dec%202009%2018%3A35%3A36%20GMT > Event-Date-Timestamp: 1262111736464194 > Event-Calling-File: sofia_reg.c > Event-Calling-Function: sofia_reg_handle_register > Event-Calling-Line-Number: 1127 > Event-Subclass: sofia%3A%3Aregister > profile-name: internal > from-user: 698 > from-host: 192.168.100.132 > presence-hosts: 192.168.100.132 > contact: %22user%22%20%3Csip%3A698%40192.168.100.130%3A5060%3E > call-id: 1909944913%40192.168.100.130 > rpid: unknown > statusd: Registered(UDP) > expires: 60 > to-user: 698 > to-host: 192.168.100.132 > network-ip: 192.168.100.130 > network-port: 5060 > username: 698 > realm: 192.168.100.132 > user-agent: UTSTARCOM%20F3000/Device%20ID-F3000_TEST > > > Am I doing something wrong, or am I missing something? > > Thanks, > Ron > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- > This email was Anti Virus checked by Astaro Security Gateway. > http://www.astaro.com From ken at ukgb.net Wed Dec 30 10:45:38 2009 From: ken at ukgb.net (Ken Gillett) Date: Wed, 30 Dec 2009 18:45:38 +0000 Subject: [Freeswitch-users] MacOSX In-Reply-To: <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> Message-ID: This is beginning to confuse me. Some say just: > - configure > - make > - make install is required, but the docs say more is needed for modules.conf. I'm still not sure if this only applies when modules.conf has been edited. Anyone help there? On 28 Dec 2009, at 14:37, Brian West wrote: > "all" is no longer needed. > > /b > > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > >> make all install sounds-install moh-install. So make install sounds-install moh-install. is required? Always? Why? Also, to bring this topic back to my original question (not that the diversity hasn't been interesting:-) How can I best compile FS on one Mac and install it onto a different Mac? Ken G i l l e t t _/_/_/_/_/_/_/_/ From jaugenstine at gmail.com Wed Dec 30 11:03:34 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Wed, 30 Dec 2009 11:03:34 -0800 Subject: [Freeswitch-users] MacOSX In-Reply-To: References: <38AF43C8-CF2F-4295-8D53-93701E056E6C@ukgb.net> <207e7a5e0912242040v3981e9a4l83045aba83e9a1f6@mail.gmail.com> <0A814262-B00D-49DB-B5EA-4AC2040B7014@freeswitch.org> Message-ID: <207e7a5e0912301103h17133fdfk22430bbce5a1718b@mail.gmail.com> Ken, configure make make install This sequence of steps builds and installs the default configuration but without the audio files. If you want the sound files installed also then: make install sounds-install moh-install Now the default sound files for conferencing, voicemail and music on hold are installed. If you want to modify the default install to customize the build you can add and remove modules in modules.conf. Then you run make/make install again to build those modules that are now included in the edited modules.conf file. Jonathan On Wed, Dec 30, 2009 at 10:45 AM, Ken Gillett wrote: > This is beginning to confuse me. Some say just: > > > - configure > > - make > > - make install > > is required, but the docs say more is needed for modules.conf. I'm still > not sure if this only applies when modules.conf has been edited. Anyone help > there? > > On 28 Dec 2009, at 14:37, Brian West wrote: > > > "all" is no longer needed. > > > > /b > > > > On Dec 25, 2009, at 3:07 AM, Ken Gillett wrote: > > > >> make all install sounds-install moh-install. > > So > > make install sounds-install moh-install. > > is required? Always? Why? > > Also, to bring this topic back to my original question (not that the > diversity hasn't been interesting:-) > > How can I best compile FS on one Mac and install it onto a different Mac? > > > > Ken G i l l e t t > > _/_/_/_/_/_/_/_/ > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/78a73fd1/attachment-0002.html From jerry.richards at teotech.com Wed Dec 30 11:16:31 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 11:16:31 -0800 Subject: [Freeswitch-users] SVN Trunk Release Label Message-ID: When you generate a release or pre-release, how can I find out what subversion trunk revision # it corresponds to? Would it make sense to make a separate subversion revision with a comment declaring the release/pre-release version (e.g. "1.0.5pre9", or "1.0.5")? Best Regards, Jerry From brian at freeswitch.org Wed Dec 30 11:49:38 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 13:49:38 -0600 Subject: [Freeswitch-users] SVN Trunk Release Label In-Reply-To: References: Message-ID: <4621070F-27B3-4D7D-B85A-D4C47F940115@freeswitch.org> You'll see an official announcement... to find the latest pre of 1.0.5 go to http://latest.freeswitch.org/ /b On Dec 30, 2009, at 1:16 PM, Jerry Richards wrote: > When you generate a release or pre-release, how can I find out what > subversion trunk revision # it corresponds to? Would it make sense to make > a separate subversion revision with a comment declaring the > release/pre-release version (e.g. "1.0.5pre9", or "1.0.5")? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Wed Dec 30 12:13:13 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 31 Dec 2009 07:13:13 +1100 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> Message-ID: <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> Hi Anthony, The only profiles I have defined are external and internal. These should be using internal... 192.168.1.120 is the FS box, which is NAT'd. Never had any problems with this being NAT'd though 192.168.1.121 is a PAP2 ATA connected to FS I don't use proxy media. I am trying to call an SPA3102, which is on the internet and NAT'd (external IP address 11.11.11.11 in the trace and internal/private ip address of 192.168.1.3). Thanks! On Thu, Dec 31, 2009 at 1:04 AM, Anthony Minessale wrote: > are you using more than one profile here? if so can you repeat the trace > with siptrace on in both profiles. > > I notice this on the trace: > > ? ?Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 > ? ?From: 1000 ;tag=70c4b62bd9443c9fo0 > > and this: > > sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000 at 192.168.1.120] > 192.168.1.120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20 > > Are you behind double nat and or is FS also behind nat? > > To address the gentleman who mentioned he was spoiled by asterisk.? Keep in > mind we have several advanced nat techniques but many of them break other > situations and they are not all turned on at once with one parameter like in > asterisk, we prefer you use the appropriate ones based on the situation. > > > > > On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith > wrote: >> >> Hi Brian, >> >> But that is all I see... I don't see any further messages after the >> 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that >> this is part of the authentication procedure, but it stops at the >> 401/Authorized message >> >> What about in the pastebin? ?Do you see anything there? ?This is when >> the call gets setup, but the IP address in the SDP says a private IP >> address ?http://pastebin.freeswitch.org/11632 >> >> On Tue, Dec 29, 2009 at 2:25 AM, Brian West wrote: >> > If you're using the 401 as an indication that it fails then you don't >> > understand how digest authentication works. ?I would have to see what >> > happens after the 401 to see if it really did fail. >> > >> > /b >> > >> > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote: >> > >> >> This is all I see and then registration fails. >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Dec 30 12:21:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 30 Dec 2009 14:21:16 -0600 Subject: [Freeswitch-users] No audio after Remote SDP: In-Reply-To: <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> References: <33c87fa30912200358h7720d50fi2f7ddd0e82b6fb1e@mail.gmail.com> <4B2E8EB4.2040105@skillsaw.com> <11343698-32EC-40C1-AC96-1EEE4C3C9DD1@freeswitch.org> <33c87fa30912201554i347e4296n936c6f9b91e9ce85@mail.gmail.com> <7ABD5B29-6E71-423C-AE8E-B55036077D4E@freeswitch.org> <33c87fa30912240316r1a908da9j82c8fc72ef3809d9@mail.gmail.com> <772A989C-0FBD-4E5B-885E-5B896075215B@freeswitch.org> <33c87fa30912300142q7a56e8f3p49031a55695cca24@mail.gmail.com> <191c3a030912300604y13421488j854a67bc18c6e926@mail.gmail.com> <33c87fa30912301213g3d24b773k86b95e7ce08f6761@mail.gmail.com> Message-ID: <7D75E6D6-8AB9-429E-A5A9-1639C0D5AD09@freeswitch.org> show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please. /b On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote: > Hi Anthony, > > The only profiles I have defined are external and internal. These > should be using internal... > > 192.168.1.120 is the FS box, which is NAT'd. Never had any problems > with this being NAT'd though > 192.168.1.121 is a PAP2 ATA connected to FS > > I don't use proxy media. > > I am trying to call an SPA3102, which is on the internet and NAT'd > (external IP address 11.11.11.11 in the trace and internal/private ip > address of 192.168.1.3). > > Thanks! From jerry.richards at teotech.com Wed Dec 30 14:47:42 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 14:47:42 -0800 Subject: [Freeswitch-users] FS Reporting Previous Presence Status, Not Current Presence Status Message-ID: I'm using two Bria softphones and I found that if I change presence a few times in succession (separated by several seconds), FS is reporting the previous status, not the current change. For example, if I change the presence from Available to Away to Available, the last PUBLISH sent from the softphone indicates Available, but then FS sends a NOTIFY with Away to the subscriber(s). Is this a bug? Best Regards, Jerry From anthony.minessale at gmail.com Wed Dec 30 15:00:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Dec 2009 17:00:51 -0600 Subject: [Freeswitch-users] FS Reporting Previous Presence Status, Not Current Presence Status In-Reply-To: References: Message-ID: <191c3a030912301500h75fa4cbeh1bb5dda249983df3@mail.gmail.com> Since its a commercial phone, we don't have it so we could not debug it, so it's hard to answer your question. On Wed, Dec 30, 2009 at 4:47 PM, Jerry Richards wrote: > I'm using two Bria softphones and I found that if I change presence a few > times in succession (separated by several seconds), FS is reporting the > previous status, not the current change. > > For example, if I change the presence from Available to Away to Available, > the last PUBLISH sent from the softphone indicates Available, but then FS > sends a NOTIFY with Away to the subscriber(s). Is this a bug? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/0eb19da5/attachment-0002.html From jerry.richards at teotech.com Wed Dec 30 15:23:49 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 30 Dec 2009 15:23:49 -0800 Subject: [Freeswitch-users] PSTN-to-Internal Call Does Not Get RoutedtoVoice Mail In-Reply-To: <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> References: <81E601F5012543C8B2BF0CC9B7EB0BF4@greyhawk.tonecommander.com><191c3a030912281530g7fd0ead8gbc463bfc98763fee@mail.gmail.com> <5CCB6773-AD61-4559-ABB2-DDDA7F84A5D7@jerris.com> Message-ID: Mike, Yes your updated driver works correctly. This is very cool. Thanks! Jerry _____ From: Michael Jerris [mailto:mike at jerris.com] Sent: Tuesday, December 29, 2009 3:37 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get RoutedtoVoice Mail try these drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.7.smg_pri.4.tgz Mike On Dec 29, 2009, at 6:17 PM, Jerry Richards wrote: I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Monday, December 28, 2009 3:31 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards wrote: Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/cd511a1d/attachment-0002.html From vmknott at gmail.com Wed Dec 30 15:24:45 2009 From: vmknott at gmail.com (VM Knott) Date: Wed, 30 Dec 2009 18:24:45 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) Message-ID: I am attempting to use multiple FreeSWITCH servers to share a common collection of voicemail boxes. Everything is working great, except for when it comes to managing the custom greetings within the individual voicemail boxes. As I understand it, when creating a custom greeting within a voicemail box, a record is stored in the voicemail_default.db file located in the ../freeswitch/db directory (which allows FS to know where to retrieve the greeting message). My original plan was to link (NFS) the remote servers to the directory to reference this file, and keep the custom greetings in an additional shared directory location (again, all of this working great so far). voicemail_prefs cols: username, domain, name_path, greeting_path, password Unfortunately, the record that is inserted into this file includes an IP address (domain) of the FS server handling the call. This complicates things if I am ?hot swapping? FS servers in and out of the server cluster. Each time I add a server to the cluster, I will have to have a process go into this file and replicate all of the records for each mailbox account, to account for the specific server ip address. Seems awkward. Am I approaching this wrong? i.e., is there a way to by pass this voicemail_prefs table? or would I have to dig into the mod_voicemail.c source code and customize it? - VMK From msc at freeswitch.org Wed Dec 30 16:02:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 16:02:16 -0800 Subject: [Freeswitch-users] Special Announcement: Latest FreeSWITCH Files Message-ID: <87f2f3b90912301602u4697b0ccjf89e6ce89d8a6fec@mail.gmail.com> Greetings all! We would like to let everyone know that we have a new place for you to download the latest FreeSWITCH source packages: http://latest.freeswitch.org. The official announcement can be read here. Thanks for all your help in making FreeSWITCH a great project and a great community. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/6d5adb95/attachment-0002.html From msc at freeswitch.org Wed Dec 30 16:23:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Dec 2009 16:23:42 -0800 Subject: [Freeswitch-users] Freeswitch 1.0.4 and Patton 4554 gateway registration problem In-Reply-To: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> References: <436A3C98-5402-4FB5-87F7-61281AFFB027@ilibris.be> Message-ID: <87f2f3b90912301623y1c0e21chf60e4f34aa78c7fa@mail.gmail.com> You definitely need to got on the latest FreeSWITCH: http://latest.freeswitch.org or use SVN. Secondly, if you are seeing this message then you've probably not got your domain configured. Is 10.156.10.93 your FreeSWITCH server? If you are using the default configs then the domain will be the IP address of your FS server. Go to the FS command line and do "eval ${domain}" and see what it says. Make sure that you don't have any strange things happening like having multiple NICs, etc. that might be wreaking havoc on your system. -MC On Wed, Dec 30, 2009 at 8:15 AM, Frederik Denkens | iLibris < fdenkens at ilibris.be> wrote: > Hi all! > > We our looking forward to replacing our legacy PBX with a Freeswitch > platform, but are struggling with getting our Patton SmartNode 4554 > (SIP/ISDN) gateway hooked up. Any assistance from the community would be > great! > > First step is getting Freeswitch and the Patton talking to each other. > > Goal? To be able to have incoming and outgoing calls going over the Patton > to the ISDN network. > > Internal PBX: > - Freeswitch 1.0.4 installed from tar with default (demo) setup on IP > 10.156.10.93 > - Patton 4554 with simple config on IP 10.156.10.90 > > We set it up that the Patton registers with the PBX and we get the error: > > --------------------- > *2009-12-30 17:07:08.11907 [WARNING] sofia_reg.c:1771 Can't find user [ > 101 at 10.156.10.93]* > *You must define a domain called '10.156.10.93' in your directory and add > a user with the id="101" attribute* > *and you must configure your device to use the proper domain in it's > authentication credentials.* > --------------------- > > Find more info below. > > So any help would be great! > > Many thanks! > > Frederik Denkens > Belgium > *+32 475 96 04 93 * > > > > > We defined a gateway in conf/sip_profiles/external/patton.xml: > --------------------- > > --> > > > > > > --------------------- > And the relevant parts of the Patton config: > --------------------- > # define auth > authentication-service AUTH_SVC > username 101 password 101 > > # patton registers > location-service LOCATION_SVC > domain 1 10.156.10.93 > identity 101 > authentication outbound > authenticate 1 authentication-service AUTH_SVC username 101 > registration outbound > registrar 10.156.10.93 5080 > lifetime 3600 > register auto > > context sip-gateway GW_SIP > interface IF_SIP > bind interface IF_IP_WAN context router port 5060 > > context sip-gateway GW_SIP > bind location-service LOCATION_SVC > no shutdown > --------------------- > Output from 'sofia status' > --------------------- > 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG > Name Type Data State > > ================================================================================================= > internal profile sip:mod_sofia at 10.156.10.93:5060 RUNNING > (0) > external profile sip:mod_sofia at 10.156.10.93:5080 RUNNING > (0) > example.com gateway sip:joeuser at example.com NOREG > 10.156.10.90 gateway sip:101 at 10.156.10.90 NOREG > 10.156.10.93 alias internal ALIASED > > ================================================================================================= > --------------------- > sip trace > --------------------- > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP 10.156.10.90:5060;branch=z9hG4bKe231fe7b97aa2c0cf > From: ;tag=b793adfadf > To: ;tag=Ua16yjtFgDQ5c > Call-ID: add8fbd86264310b > CSeq: 12517 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > *Before printing this e-mail, please consider the impact on the > environment.* > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091230/67ad02ee/attachment-0002.html From freeswitch at aastral.net Wed Dec 30 20:06:09 2009 From: freeswitch at aastral.net (Bill W.) Date: Wed, 30 Dec 2009 23:06:09 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <4B3C2331.4030504@aastral.net> Hey VM, Couldn't you just have your core use ODBC instead of SQLite? http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Bill W VM Knott wrote: > I am attempting to use multiple FreeSWITCH servers to share a common > collection of voicemail boxes. Everything is working great, except > for when it comes to managing the custom greetings within the > individual voicemail boxes. > > As I understand it, when creating a custom greeting within a voicemail > box, a record is stored in the voicemail_default.db file located in > the ../freeswitch/db directory (which allows FS to know where to > retrieve the greeting message). My original plan was to link (NFS) > the remote servers to the directory to reference this file, and keep > the custom greetings in an additional shared directory location > (again, all of this working great so far). > > voicemail_prefs cols: username, domain, name_path, greeting_path, password > > Unfortunately, the record that is inserted into this file includes an > IP address (domain) of the FS server handling the call. This > complicates things if I am ?hot swapping? FS servers in and out of the > server cluster. Each time I add a server to the cluster, I will have > to have a process go into this file and replicate all of the records > for each mailbox account, to account for the specific server ip > address. Seems awkward. > > Am I approaching this wrong? i.e., is there a way to by pass this > voicemail_prefs table? > or would I have to dig into the mod_voicemail.c source code and customize it? > > - VMK > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at aastral.net Wed Dec 30 21:53:58 2009 From: freeswitch at aastral.net (Bill W.) Date: Thu, 31 Dec 2009 00:53:58 -0500 Subject: [Freeswitch-users] Sofia sqlite error Message-ID: <4B3C3C76.10007@aastral.net> Hi All I have 3 sofia profiles. Two of them are using sqlite, and one is using odbc. Whenever I restart a profile that is using sqlite, I get the following error: 2009-12-31 00:29:56.914495 [ERR] switch_core_db.c:109 SQL ERR [table sip_shared_appearance_dialogs has no column named network_ip] I'm on version FreeSWITCH Version 1.0.trunk (16055) I did a checkout from trunk and switch_core_db.c hasn't changed from the version I have, so I didn't bother to recompile. Thoughts? Thanks! Bill From pmhshz at gmail.com Wed Dec 30 22:49:48 2009 From: pmhshz at gmail.com (shehzad p) Date: Wed, 30 Dec 2009 22:49:48 -0800 (PST) Subject: [Freeswitch-users] Re cording call into existing file Message-ID: <26975973.post@talk.nabble.com> Hi, while recording a file using session_record, can i continue the existing recorded file? So that the existing record will remain as it is and new recording will be added into that file? Thanks msp -- View this message in context: http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tzury.by at reguluslabs.com Wed Dec 30 22:59:30 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 31 Dec 2009 08:59:30 +0200 Subject: [Freeswitch-users] Configure FS to reply with domain instead of with IP to: user@domain instead of to: user@ip:port Message-ID: <10128ef10912302259o60b989ecxd2a5958a24d98b2c@mail.gmail.com> Hi all, I would like to know what should be done in order to have FreeSWITCH reply with domains instead of IP-Addresses. In my server, DNS NAPTR & SRV Records are all set and working fine. Clients are connected using our pjsip port for windows mobile where transport=tls. When we connect to a public SIP service (e.g. iptel.org) we get the domain when performing INVITE request. We want to gain this behavior in FS as well so it will reply similarly. I am referring to the "TO" part in the message e.g. To: sip:user at domain instead of To: see below our client log in both cases (iptel and FS) thanks in advance for your help, Tzury ## IPTel incoming invite recording DEBUG: 01:26:01.088 pjsua_core.c RX 1746 bytes Request msg INVITE/cseq=7942 (rdata0046B234) from UDP 213.192.59.75:5060: INVITE sip:tal2 at 95.35.38.192:5060;transport=UDP SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 213.192.59.75;branch=z9hG4bKd3a7.502cfbb.0;i=12c8 Via: SIP/2.0/tcp 10.0.0.103:3600;received=80.74.97.189;rport=3600;branch=z9hG4bKPj868b3031276d4f0d9f2338b854b63191 Max-Forwards: 16 From: sip:Tal3 at iptel.org;tag=ff25112de1c843a58df49f57c54a29a2 To: sip:tal2 at iptel.org Contact: ### Freeswitch incoming invite recording DEBUG: 01:28:39.163 pjsua_core.c RX 1257 bytes Request msg INVITE/cseq=119747894 (rdata004B8384) from UDP 67.23.5.142:5060: INVITE sip:1002 at 95.35.115.158:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKyme7e23ggp2QB Max-Forwards: 69 From: "Extension 1000" ;tag=6F8v2e9NNgZ6a To: Call-ID: a9e42c07-10e5-122d-de9e-40402384297d CSeq: 119747894 INVITE Contact: From jcasale at activenetwerx.com Thu Dec 31 00:17:28 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 08:17:28 +0000 Subject: [Freeswitch-users] Outbound Dial Configuration Message-ID: I am starting to migrate an Asterisk box over with a tdm card w/ 1 fxo port this office uses for redundancy when either their sip provider or net connection drops. In Asterisk, I had a long macro for attempting the sip providers first, then finally getting to the dahdi line. Can I simply do the following: to attempt my sip provider first always, then hit span 1/port 1 of my tdm card if it's the provider is not available? Is this an elegant enough way to do this for an office of < 10 phones? Thanks, jlc From jcasale at activenetwerx.com Thu Dec 31 00:45:27 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 08:45:27 +0000 Subject: [Freeswitch-users] Zap problems Message-ID: I am trying to setup a TDM410P card nuder freeswitch-1.0.5-20091230-0400 and fs is reporting it as down? /etc/dahdi/system.conf as it was under Asterisk, so its correct (set to fxs). /opt/freeswitch/conf/zt.conf is default. # cat /opt/freeswitch/conf/openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 # cat /opt/freeswitch/conf/autoload_configs/openzap.conf.xml # dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM410P Board 1 name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM410P location=PCI Bus 10 Slot 02 basechan=1 totchans=4 irq=233 type=analog port=1,FXO port=2,none port=3,none port=4,none Yet fs_cli reports: freeswitch at 127.0.0.1@internal> oz list +OK span: 1 (span1) type: analog chan_count: 1 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options 3way freeswitch at 127.0.0.1@internal> oz dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 type: FXO state: DOWN last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE Any idea what I am missing? The udev rules are set to the user/group fs runs under. Starting fs at the cli shows the following output, which looks fine? 2009-12-31 01:39:17.592736 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/openzap.conf. 2009-12-31 01:39:17.592803 [DEBUG] zap_io.c:2362 found config for span 2009-12-31 01:39:17.593066 [NOTICE] ozmod_zt.c:1166 Using DAHDI control device 2009-12-31 01:39:17.593112 [INFO] zap_io.c:2579 Loading IO from /opt/freeswitch/mod/ozmod_zt.so [zt] 2009-12-31 01:39:17.593145 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/zt.conf. 2009-12-31 01:39:17.593258 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000 2009-12-31 01:39:17.593298 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000 2009-12-31 01:39:17.593353 [INFO] zap_io.c:2379 auto-loaded 'zt' 2009-12-31 01:39:17.593399 [DEBUG] zap_io.c:2400 created span 1 (span1) of type zt 2009-12-31 01:39:17.593426 [DEBUG] zap_io.c:2413 span 1 [name]=[OpenZAP] 2009-12-31 01:39:17.593457 [DEBUG] zap_io.c:2413 span 1 [number]=[1] 2009-12-31 01:39:17.593485 [DEBUG] zap_io.c:2413 span 1 [fxo-channel]=[1] 2009-12-31 01:39:17.593513 [DEBUG] zap_io.c:2442 setting trunk type to 'FXO' start(KEWL) 2009-12-31 01:39:17.593593 [INFO] ozmod_zt.c:385 configuring device /dev/dahdi/channel channel 1 as OpenZAP device 1:1 fd:35 2009-12-31 01:39:17.593657 [INFO] zap_io.c:2502 Configured 1 channel(s) although stopping fs yielded this: 2009-12-31 01:42:44.545768 [INFO] zap_io.c:269 Closing channel zt:1:1 fd:35 2009-12-31 01:42:44.574577 [ERR] ozmod_analog.c:953 Failure Polling event! [no matching descriptor] 2009-12-31 01:42:45.546544 [INFO] zap_io.c:2694 Unloading /opt/freeswitch/mod/ozmod_analog.so 2009-12-31 01:42:45.546735 [INFO] zap_io.c:2679 Unloading IO zt 2009-12-31 01:42:45.546756 [INFO] zap_io.c:2694 Unloading /opt/freeswitch/mod/ozmod_zt.so Thanks! jlc From codecomplete at free.fr Thu Dec 31 04:52:14 2009 From: codecomplete at free.fr (Fred-145) Date: Thu, 31 Dec 2009 04:52:14 -0800 (PST) Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> References: <26808383.post@talk.nabble.com> <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> Message-ID: <26977788.post@talk.nabble.com> Hristo Benev-3 wrote: > Just for your information there is a version of nmap for windows. So you > can do the test from your desktop. Thanks for the info but I'd like to run a test from the Net to see if the NAT firewall did open the required ports for SIP/RTP. I don't have a second PC elsewhere on the Net where I could aim nmap at my home ADSL modem/router. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p26977788.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Thu Dec 31 06:14:34 2009 From: jaybinks at gmail.com (Jay Binks) Date: Fri, 1 Jan 2010 00:14:34 +1000 Subject: [Freeswitch-users] Happy new year all Message-ID: Happy new years... 2010 a great new year for FS, looking forward to cluecon already ;) Jay From anthony.minessale at gmail.com Thu Dec 31 07:04:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:04:17 -0600 Subject: [Freeswitch-users] Re cording call into existing file In-Reply-To: <26975973.post@talk.nabble.com> References: <26975973.post@talk.nabble.com> Message-ID: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> set RECORD_APPEND=true on the channel and all recordings will behave this way to formats which support it (curently mod_sndfile for WAV etc) On Thu, Dec 31, 2009 at 12:49 AM, shehzad p wrote: > > Hi, > > while recording a file using session_record, can i continue the existing > recorded file? So that the existing record will remain as it is and new > recording will be added into that file? > > Thanks > msp > -- > View this message in context: > http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/72f665c9/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 31 07:07:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:07:04 -0600 Subject: [Freeswitch-users] Happy new year all In-Reply-To: References: Message-ID: <191c3a030912310707q2f72b63dg10f956ecc54df19f@mail.gmail.com> Thanks for the head's up! It's nice to have eyes into the future. Especially 10 years ago in the eve of y2k =p On Thu, Dec 31, 2009 at 8:14 AM, Jay Binks wrote: > Happy new years... > > 2010 a great new year for FS, looking forward to cluecon already ;) > > Jay > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/5ce8d3d9/attachment-0002.html From rupa at rupa.com Thu Dec 31 07:07:58 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 31 Dec 2009 09:07:58 -0600 Subject: [Freeswitch-users] Outbound Dial Configuration In-Reply-To: References: Message-ID: Yes. On Thu, Dec 31, 2009 at 2:17 AM, Joseph L. Casale wrote: > I am starting to migrate an Asterisk box over with a tdm card w/ 1 fxo port this > office uses for redundancy when either their sip provider or net connection drops. > > In Asterisk, I had a long macro for attempting the sip providers first, then finally > getting to the dahdi line. > > Can I simply do the following: > > > > to attempt my sip provider first always, then hit span 1/port 1 of my tdm card if it's > the provider is not available? Is this an elegant enough way to do this for an office > of < 10 phones? > > Thanks, > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From saigop at gmail.com Thu Dec 31 07:11:14 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 31 Dec 2009 20:41:14 +0530 Subject: [Freeswitch-users] Module for TTS Message-ID: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> Hi, I would like to develop a module for TTS using http://sourceforge.net/projects/dhvani/ for Freeswitch, How to start with it? -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/eef48098/attachment-0002.html From brian at freeswitch.org Thu Dec 31 07:21:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 31 Dec 2009 09:21:06 -0600 Subject: [Freeswitch-users] Module for TTS In-Reply-To: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> References: <2ea4d47e0912310711s5f7eadc6q4d386dc559e35d89@mail.gmail.com> Message-ID: You might want to take a look at mod_tts_commandline.c as that tts engine is GPL which is license incompatible with FreeSWITH unless they provide an exception. Thanks, Brian On Dec 31, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > Hi, > > I would like to develop a module for TTS using http://sourceforge.net/projects/dhvani/ for Freeswitch, How to start with it? > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/3896a974/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 31 07:32:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 09:32:29 -0600 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <4B3C3C76.10007@aastral.net> References: <4B3C3C76.10007@aastral.net> Message-ID: <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> delete all the files in your /usr/local/freeswitch/db and rebuild trunk and try again. On Wed, Dec 30, 2009 at 11:53 PM, Bill W. wrote: > Hi All > > I have 3 sofia profiles. Two of them are using sqlite, and one is using > odbc. > > Whenever I restart a profile that is using sqlite, I get the following > error: > > 2009-12-31 00:29:56.914495 [ERR] switch_core_db.c:109 SQL ERR [table > sip_shared_appearance_dialogs has no column named network_ip] > > I'm on version FreeSWITCH Version 1.0.trunk (16055) > > I did a checkout from trunk and switch_core_db.c hasn't changed from the > version I have, so I didn't bother to recompile. > > Thoughts? > > Thanks! > Bill > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/c4b9338b/attachment-0002.html From freeswitch at peely.com Thu Dec 31 07:49:50 2009 From: freeswitch at peely.com (peely) Date: Thu, 31 Dec 2009 07:49:50 -0800 (PST) Subject: [Freeswitch-users] Call through gateway without register > sends to gateway name? Message-ID: <26979541.post@talk.nabble.com> Hi, I have a problem where I'm trying to send calls to a gateway that does not support registration. In my external sip profile directory I have a file containing: Then in my dialplan I have a dialplan with an action of: However, when I call out, the Sofia diag shows: sres_send_dns_query(0x7f19ac011150, 0x7f19a401cdf0) id=20898 NAPTR mygateway (to [172.16.1.1]:53) Could somebody please tell me how I get the gateway config to send INVITEs to a specific IP? Thanks, Neil. -- View this message in context: http://old.nabble.com/Call-through-gateway-without-register-%3E-sends-to-gateway-name--tp26979541p26979541.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Dec 31 08:00:05 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 31 Dec 2009 08:00:05 -0800 Subject: [Freeswitch-users] Zap problems In-Reply-To: References: Message-ID: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> Actually DOWN simply means idle. Go ahead and make some calls - your config looks okay. -MC On Dec 31, 2009, at 12:45 AM, "Joseph L. Casale" wrote: > I am trying to setup a TDM410P card nuder > freeswitch-1.0.5-20091230-0400 > and fs is reporting it as down? > > /etc/dahdi/system.conf as it was under Asterisk, so its correct (set > to fxs). > /opt/freeswitch/conf/zt.conf is default. > > # cat /opt/freeswitch/conf/openzap.conf > [span zt] > name => OpenZAP > number => 1 > fxo-channel => 1 > > # cat /opt/freeswitch/conf/autoload_configs/openzap.conf.xml > > > > > > > > > > > > > > > > > > > > > # dahdi_scan > [1] > active=yes > alarms=OK > description=Wildcard TDM410P Board 1 > name=WCTDM/0 > manufacturer=Digium > devicetype=Wildcard TDM410P > location=PCI Bus 10 Slot 02 > basechan=1 > totchans=4 > irq=233 > type=analog > port=1,FXO > port=2,none > port=3,none > port=4,none > > Yet fs_cli reports: > > freeswitch at 127.0.0.1@internal> oz list > +OK > span: 1 (span1) > type: analog > chan_count: 1 > dialplan: XML > context: default > dial_regex: > fail_dial_regex: > hold_music: > analog_options 3way > > freeswitch at 127.0.0.1@internal> oz dump 1 1 > span_id: 1 > chan_id: 1 > physical_span_id: 1 > physical_chan_id: 1 > type: FXO > state: DOWN > last_state: DOWN > cid_date: > cid_name: > cid_num: > ani: > aniII: > dnis: > rdnis: > cause: NONE > > Any idea what I am missing? The udev rules are set to the user/group > fs runs under. > Starting fs at the cli shows the following output, which looks fine? > > 2009-12-31 01:39:17.592736 [DEBUG] zap_config.c:56 Configuration > file is /opt/freeswitch/conf/openzap.conf. > 2009-12-31 01:39:17.592803 [DEBUG] zap_io.c:2362 found config for span > 2009-12-31 01:39:17.593066 [NOTICE] ozmod_zt.c:1166 Using DAHDI > control device > 2009-12-31 01:39:17.593112 [INFO] zap_io.c:2579 Loading IO from /opt/ > freeswitch/mod/ozmod_zt.so [zt] > 2009-12-31 01:39:17.593145 [DEBUG] zap_config.c:56 Configuration > file is /opt/freeswitch/conf/zt.conf. > 2009-12-31 01:39:17.593258 [INFO] ozmod_zt.c:556 Setting rxgain val > to 0.000000 > 2009-12-31 01:39:17.593298 [INFO] ozmod_zt.c:565 Setting txgain val > to 0.000000 > 2009-12-31 01:39:17.593353 [INFO] zap_io.c:2379 auto-loaded 'zt' > 2009-12-31 01:39:17.593399 [DEBUG] zap_io.c:2400 created span 1 > (span1) of type zt > 2009-12-31 01:39:17.593426 [DEBUG] zap_io.c:2413 span 1 [name]= > [OpenZAP] > 2009-12-31 01:39:17.593457 [DEBUG] zap_io.c:2413 span 1 [number]=[1] > 2009-12-31 01:39:17.593485 [DEBUG] zap_io.c:2413 span 1 [fxo-channel] > =[1] > 2009-12-31 01:39:17.593513 [DEBUG] zap_io.c:2442 setting trunk type > to 'FXO' start(KEWL) > 2009-12-31 01:39:17.593593 [INFO] ozmod_zt.c:385 configuring device / > dev/dahdi/channel channel 1 as OpenZAP device 1:1 fd:35 > 2009-12-31 01:39:17.593657 [INFO] zap_io.c:2502 Configured 1 channel > (s) > > although stopping fs yielded this: > > 2009-12-31 01:42:44.545768 [INFO] zap_io.c:269 Closing channel zt: > 1:1 fd:35 > 2009-12-31 01:42:44.574577 [ERR] ozmod_analog.c:953 Failure Polling > event! [no matching descriptor] > 2009-12-31 01:42:45.546544 [INFO] zap_io.c:2694 Unloading /opt/ > freeswitch/mod/ozmod_analog.so > 2009-12-31 01:42:45.546735 [INFO] zap_io.c:2679 Unloading IO zt > 2009-12-31 01:42:45.546756 [INFO] zap_io.c:2694 Unloading /opt/ > freeswitch/mod/ozmod_zt.so > > Thanks! > jlc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Dec 31 08:07:19 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 31 Dec 2009 10:07:19 -0600 Subject: [Freeswitch-users] Call through gateway without register > sends to gateway name? In-Reply-To: <26979541.post@talk.nabble.com> References: <26979541.post@talk.nabble.com> Message-ID: <04A52674-4ABB-4EAB-BFA9-65A18C3700C5@freeswitch.org> If you don't call your gateway the dns name or the IP then you'll have to specify the proxy/username/password/from-domain settings. /b On Dec 31, 2009, at 9:49 AM, peely wrote: > > > From jcasale at activenetwerx.com Thu Dec 31 08:16:12 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 16:16:12 +0000 Subject: [Freeswitch-users] Zap problems In-Reply-To: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> References: <65045AA1-5C1A-4CAD-9E52-4EB7E16A4DD9@freeswitch.org> Message-ID: >Actually DOWN simply means idle. Go ahead and make some calls - your >config looks okay. Heh, I had the guys onsite re-install the old discs as I figured I couldn't get the thing operational by start of business... I'll be back on it once I recover from tonight:) Thanks for all the help guys, happy new year! jlc From jcasale at activenetwerx.com Thu Dec 31 08:19:04 2009 From: jcasale at activenetwerx.com (Joseph L. Casale) Date: Thu, 31 Dec 2009 16:19:04 +0000 Subject: [Freeswitch-users] Scanning my firewall for open UDP ports? In-Reply-To: <26977788.post@talk.nabble.com> References: <26808383.post@talk.nabble.com> <1725851358.229612.1261055690205.JavaMail.apache@mail21.abv.bg> <26977788.post@talk.nabble.com> Message-ID: >Thanks for the info but I'd like to run a test from the Net to see if the >NAT firewall did open the required ports for SIP/RTP. I don't have a second >PC elsewhere on the Net where I could aim nmap at my home ADSL modem/router. http://www.t1shopper.com/tools/port-scanner/ One of the few that allow you to scan an ip that's not yours... From msc at freeswitch.org Thu Dec 31 09:20:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Dec 2009 09:20:14 -0800 Subject: [Freeswitch-users] freeswitch and H323 In-Reply-To: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> References: <7aa8bd9d0912300514r68c90b12u7c631a649981cfa3@mail.gmail.com> Message-ID: <87f2f3b90912310920q1821fe0eu5fd74e553e4cb12@mail.gmail.com> Are you trying to use mod_h323 or mod_opal? They are both works in progress, but the latter is farther along than the former. Use the latest FreeSWITCH trunk (or latest.freeswitch.org) and run the buildopal.sh script in the build directory. If you have any build issues then paste the log on pastebin.freeswitch.org and reply to this thread with the PB URL so that we can take a look. -MC On Wed, Dec 30, 2009 at 5:14 AM, Pete Kay wrote: > Hi, > > has anyone been able to get H323 to work? > > I have problem trying to get it compiled with either 1.0.4 or 1.0.5. > > Thanks, > pete > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/b8ee3170/attachment-0002.html From msc at freeswitch.org Thu Dec 31 09:31:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 Dec 2009 09:31:07 -0800 Subject: [Freeswitch-users] Can FS work with an external packet relay such as rtpproxy? In-Reply-To: <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> References: <6e40a4420912300619x43afa97ci9d57424f0e378850@mail.gmail.com> <5F994EB0-408B-49B5-A6D1-6DECC3F36D56@freeswitch.org> Message-ID: <87f2f3b90912310931o35387aceq6c7f9cca7726bdbf@mail.gmail.com> On Wed, Dec 30, 2009 at 8:23 AM, Brian West wrote: > Someone could pay to have support added but as it stands we don't have it > yet. If its going to be done it has to be proper mgcp support not half ass > support. > > As Brian noted on yesterday's conf call he occasionally sends "terse" emails. However, being terse frequently gets the point across in no uncertain terms. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/bc43cfab/attachment-0002.html From freeswitch at aastral.net Thu Dec 31 11:07:34 2009 From: freeswitch at aastral.net (Bill W) Date: Thu, 31 Dec 2009 14:07:34 -0500 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> References: <4B3C3C76.10007@aastral.net> <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> Message-ID: <4B3CF676.8040908@aastral.net> Worked! Was that a one-time thing or do I need to blow away the sqlite database every time I upgrade? Or more generally, how does FreeSWITCH handle database updates and what should the procedures be to ensure the database remains current and no data is lost by having to remove database files? Thanks! Bill Anthony Minessale wrote: > delete all the files in your /usr/local/freeswitch/db and rebuild trunk > and try again. > > From anthony.minessale at gmail.com Thu Dec 31 12:02:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 14:02:45 -0600 Subject: [Freeswitch-users] Sofia sqlite error In-Reply-To: <4B3CF676.8040908@aastral.net> References: <4B3C3C76.10007@aastral.net> <191c3a030912310732h67f0b4f3o2f4917250e108fe0@mail.gmail.com> <4B3CF676.8040908@aastral.net> Message-ID: <191c3a030912311202t15d1fa5p4799de5114bb867c@mail.gmail.com> usually it does a test and erases the table and recreates it when need be. On Thu, Dec 31, 2009 at 1:07 PM, Bill W wrote: > Worked! > > Was that a one-time thing or do I need to blow away the sqlite database > every time I upgrade? > > Or more generally, how does FreeSWITCH handle database updates and what > should the procedures be to ensure the database remains current and no > data is lost by having to remove database files? > > Thanks! > Bill > > > Anthony Minessale wrote: > > delete all the files in your /usr/local/freeswitch/db and rebuild trunk > > and try again. > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/37212650/attachment-0002.html From stevendt at primrosebank.net Thu Dec 31 12:22:06 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 31 Dec 2009 20:22:06 -0000 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does it mean ? Message-ID: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> Hi Guys Happy New Year ! - 4 hours to midnight here - what am I doing looking at the console log ???? Anyway, all the best for 2010, and when someone gets a chance to look at this, could someone please help me understand what the following error message means ? I've seen it a few times before and not seen anything untoward. My "production" system is running 1.0.4 - reported as (14460) but it is somewhat later than that (15xxx). I plan on doing a new SVN build over the holiday, but don't expect this error to go away ? Message is along the lines . . . . date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" regards Dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/87c358d9/attachment-0002.html From anthony.minessale at gmail.com Thu Dec 31 12:33:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 14:33:46 -0600 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does it mean ? In-Reply-To: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> Message-ID: <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> It means you have a profound jump in time like someone set the clock to an earlier time and it's recovering by re-syncing the clock. You can avoid this by using a monotonic clock which must not be possible on your system because it uses it by default. What OS is it? On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson wrote: > Hi Guys > > Happy New Year ! > > - 4 hours to midnight here - what am I doing looking at the console log > ???? > > > Anyway, all the best for 2010, and when someone gets a chance to look at > this, could someone please help me understand what the following error > message means ? > > I've seen it a few times before and not seen anything untoward. My > "production" system is running 1.0.4 - reported as (14460) but it is > somewhat later than that (15xxx). I plan on doing a new SVN build over the > holiday, but don't expect this error to go away ? > > Message is along the lines . . . . > > > date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" > > regards > Dave > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/f10f4972/attachment-0002.html From vmknott at gmail.com Thu Dec 31 14:03:57 2009 From: vmknott at gmail.com (VM Knott) Date: Thu, 31 Dec 2009 17:03:57 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) Message-ID: Thank you Bill for the input, but changing how I access the data source does not solve my problem. I was hoping to avoid the management of IP Addresses for every voicemail box on the system. Is there a way for me to set a default greeting to all voicemail boxes globally, without having to go to a repository (regardless of means of access) for each mailbox? ---------- Forwarded message ---------- From: "Bill W." To: freeswitch-users at lists.freeswitch.org Date: Wed, 30 Dec 2009 23:06:09 -0500 Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) Hey VM, Couldn't you just have your core use ODBC instead of SQLite? http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Bill W VM Knott wrote: > I am attempting to use multiple FreeSWITCH servers to share a common > collection of voicemail boxes. Everything is working great, except > for when it comes to managing the custom greetings within the > individual voicemail boxes. > > As I understand it, when creating a custom greeting within a voicemail > box, a record is stored in the voicemail_default.db file located in > the ../freeswitch/db directory (which allows FS to know where to > retrieve the greeting message). My original plan was to link (NFS) > the remote servers to the directory to reference this file, and keep > the custom greetings in an additional shared directory location > (again, all of this working great so far). > > voicemail_prefs cols: username, domain, name_path, greeting_path, password > > Unfortunately, the record that is inserted into this file includes an > IP address (domain) of the FS server handling the call. This > complicates things if I am ?hot swapping? FS servers in and out of the > server cluster. Each time I add a server to the cluster, I will have > to have a process go into this file and replicate all of the records > for each mailbox account, to account for the specific server ip > address. Seems awkward. > > Am I approaching this wrong? i.e., is there a way to by pass this > voicemail_prefs table? > or would I have to dig into the mod_voicemail.c source code and customize it? > > - VMK > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Dec 31 14:19:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Dec 2009 16:19:03 -0600 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <191c3a030912311419h3540a684q2f259aa8687f04a4@mail.gmail.com> use a real domain instead of IP might work. On Thu, Dec 31, 2009 at 4:03 PM, VM Knott wrote: > Thank you Bill for the input, but changing how I access the data > source does not solve my problem. > I was hoping to avoid the management of IP Addresses for every > voicemail box on the system. > > Is there a way for me to set a default greeting to all voicemail boxes > globally, without having to go to a repository (regardless of means of > access) for each mailbox? > > > > ---------- Forwarded message ---------- > From: "Bill W." > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 30 Dec 2009 23:06:09 -0500 > Subject: Re: [Freeswitch-users] Voicemail Question (using multiple servers) > Hey VM, > > Couldn't you just have your core use ODBC instead of SQLite? > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > -Bill W > > > VM Knott wrote: > > I am attempting to use multiple FreeSWITCH servers to share a common > > collection of voicemail boxes. Everything is working great, except > > for when it comes to managing the custom greetings within the > > individual voicemail boxes. > > > > As I understand it, when creating a custom greeting within a voicemail > > box, a record is stored in the voicemail_default.db file located in > > the ../freeswitch/db directory (which allows FS to know where to > > retrieve the greeting message). My original plan was to link (NFS) > > the remote servers to the directory to reference this file, and keep > > the custom greetings in an additional shared directory location > > (again, all of this working great so far). > > > > voicemail_prefs cols: username, domain, name_path, greeting_path, > password > > > > Unfortunately, the record that is inserted into this file includes an > > IP address (domain) of the FS server handling the call. This > > complicates things if I am ?hot swapping? FS servers in and out of the > > server cluster. Each time I add a server to the cluster, I will have > > to have a process go into this file and replicate all of the records > > for each mailbox account, to account for the specific server ip > > address. Seems awkward. > > > > Am I approaching this wrong? i.e., is there a way to by pass this > > voicemail_prefs table? > > or would I have to dig into the mod_voicemail.c source code and customize > it? > > > > - VMK > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/94d6d7e1/attachment-0002.html From stevendt at primrosebank.net Thu Dec 31 15:02:23 2009 From: stevendt at primrosebank.net (Dave Stevenson) Date: Thu, 31 Dec 2009 23:02:23 -0000 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> Message-ID: Hi Anthony it's Windows XP - there was no time change on the PC any time close to this event - or indeed for weeks since FreeSwitch was installed. I'll check, but I don't think the machine is even using SNTP. What does FreeSwitch define as a "profound jump in time" ? regards Dave (off to see the New Year in now !) ----- Original Message ----- From: Anthony Minessale To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 31, 2009 8:33 PM Subject: Re: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? It means you have a profound jump in time like someone set the clock to an earlier time and it's recovering by re-syncing the clock. You can avoid this by using a monotonic clock which must not be possible on your system because it uses it by default. What OS is it? On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson wrote: Hi Guys Happy New Year ! - 4 hours to midnight here - what am I doing looking at the console log ???? Anyway, all the best for 2010, and when someone gets a chance to look at this, could someone please help me understand what the following error message means ? I've seen it a few times before and not seen anything untoward. My "production" system is running 1.0.4 - reported as (14460) but it is somewhat later than that (15xxx). I plan on doing a new SVN build over the holiday, but don't expect this error to go away ? Message is along the lines . . . . date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" regards Dave _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/655bb922/attachment-0002.html From rupa at rupa.com Thu Dec 31 15:56:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 31 Dec 2009 17:56:24 -0600 Subject: [Freeswitch-users] "Reverse Clock Skew Detected" - what does itmean ? In-Reply-To: References: <9512D13B624C44E8A09EFC22436D7EC9@bp1.ad.bp.com> <191c3a030912311233i68a2126fge920867a4608ab3d@mail.gmail.com> Message-ID: Don't most modern windows operating systems automatically sync with time.windows.com or something like that? On Thu, Dec 31, 2009 at 5:02 PM, Dave Stevenson wrote: > Hi Anthony > > it's Windows XP - there was no time change on the PC any time close to this > event - or indeed for weeks since FreeSwitch was installed. > > I'll check, but I don't think the machine is even using SNTP. > > What does FreeSwitch define as a "profound jump in time" ? > > regards > Dave > > (off to see the New Year in now !) > > ----- Original Message ----- > *From:* Anthony Minessale > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, December 31, 2009 8:33 PM > *Subject:* Re: [Freeswitch-users] "Reverse Clock Skew Detected" - what > does itmean ? > > It means you have a profound jump in time like someone set the clock to an > earlier time and it's recovering by re-syncing the clock. > You can avoid this by using a monotonic clock which must not be possible on > your system because it uses it by default. > > What OS is it? > > > On Thu, Dec 31, 2009 at 2:22 PM, Dave Stevenson > wrote: > >> Hi Guys >> >> Happy New Year ! >> >> - 4 hours to midnight here - what am I doing looking at the console log >> ???? >> >> >> Anyway, all the best for 2010, and when someone gets a chance to look at >> this, could someone please help me understand what the following error >> message means ? >> >> I've seen it a few times before and not seen anything untoward. My >> "production" system is running 1.0.4 - reported as (14460) but it is >> somewhat later than that (15xxx). I plan on doing a new SVN build over the >> holiday, but don't expect this error to go away ? >> >> Message is along the lines . . . . >> >> >> date & time "[CRIT] switch_time.c:454 Reverse Clock Skew Detected!" >> >> regards >> Dave >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091231/9e3e159a/attachment-0002.html From sharad at coraltele.com Thu Dec 31 22:20:47 2009 From: sharad at coraltele.com (Sharad) Date: Thu, 31 Dec 2009 22:20:47 -0800 (PST) Subject: [Freeswitch-users] Self alarm In-Reply-To: <1262250725607-4235713.post@n2.nabble.com> References: <1262250725607-4235713.post@n2.nabble.com> Message-ID: <1262326847726-4238924.post@n2.nabble.com> Hi I am also intresting in the same. Is there any script for this functionality. Regards -- View this message in context: http://n2.nabble.com/Self-alarm-tp4235713p4238924.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at aastral.net Thu Dec 31 22:31:14 2009 From: freeswitch at aastral.net (Bill W.) Date: Fri, 01 Jan 2010 01:31:14 -0500 Subject: [Freeswitch-users] Voicemail Question (using multiple servers) In-Reply-To: References: Message-ID: <4B3D96B2.4060402@aastral.net> So is the problem that you're having to replicate the voicemail database across switches in the cluster or is the problem the content of the entries in voicemail database? Because in your original post you're speaking of trying to share the voicemail db over NFS. Thanks, Bill VM Knott wrote: > Thank you Bill for the input, but changing how I access the data > source does not solve my problem. > I was hoping to avoid the management of IP Addresses for every > voicemail box on the system. > > Is there a way for me to set a default greeting to all voicemail boxes > globally, without having to go to a repository (regardless of means of > access) for each mailbox? > > > From freeswitch at aastral.net Thu Dec 31 23:29:46 2009 From: freeswitch at aastral.net (Bill W.) Date: Fri, 01 Jan 2010 02:29:46 -0500 Subject: [Freeswitch-users] Re cording call into existing file In-Reply-To: <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> References: <26975973.post@talk.nabble.com> <191c3a030912310704g5c57d296pbbe553aa46c63c4e@mail.gmail.com> Message-ID: <4B3DA46A.4010906@aastral.net> Added to the Wiki. http://wiki.freeswitch.org/wiki/Variable_RECORD_APPEND Anthony Minessale wrote: > set RECORD_APPEND=true on the channel and all recordings will behave > this way to formats which support it > (curently mod_sndfile for WAV etc) >